| 00:01.37 | *** join/#asterisk hardwire (n=hardwire@89-208-58-66.gci.net) |
| 00:01.45 | hardwire | I remembered my nickserv password! |
| 00:01.51 | aptura | I have been told that the soundpoint series cannot pass there rtp traffice though a router because of some issues. I have not tested it personally just wanted to verify what I read. |
| 00:02.53 | benjk | just answer the question |
| 00:05.02 | *** join/#asterisk adamowitz (n=adamowit@ip68-9-201-27.ri.ri.cox.net) |
| 00:05.54 | *** join/#asterisk BhaalWK (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 00:15.24 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 00:16.00 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 00:21.49 | CtRiX | priv => sipregistration,0,IAX2,priv:${SECRET}@10.10.9.3/${NUMBER},nopartial |
| 00:22.01 | CtRiX | has anyone ever had this problem ? |
| 00:22.09 | CtRiX | test*CLI> dundi lookup 523@priv |
| 00:22.09 | CtRiX | <PROTECTED> |
| 00:22.09 | CtRiX | <PROTECTED> |
| 00:22.32 | CtRiX | the response does not contains ${NUMBER } and secret |
| 00:23.20 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:23.41 | *** part/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 00:26.24 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 00:31.42 | gambolputty | Hi. Anyone gotten * to work with Jabber? |
| 00:31.56 | jebba | heh |
| 00:32.03 | jebba | gambolputty, that's what i'm doing right now. Works fine. :) |
| 00:32.15 | jebba | i was just trying to get it to join conferences though.... |
| 00:39.06 | jebba | I need it to do that so I can send messages to a jabber2irc gateway... like this (but doesn't work): |
| 00:39.13 | jebba | exten => 1,n,JabberSend(asterisk,foo%irc.freenode.net@irc.jabber-gateway.org,pingo pongo) |
| 00:39.48 | *** join/#asterisk MRH2 (n=chatzill@host-84-9-253-120.bulldogdsl.com) |
| 00:40.19 | *** join/#asterisk bobesponja (n=pat@bas75-1-81-57-4-105.fbx.proxad.net) |
| 00:41.04 | MRH2 | hi anyone know if that recent commit to chan_sip regarding polycom subscribe requests would effect reinvites at all? |
| 00:43.58 | MRH2 | this is the one i mean - http://lists.digium.com/pipermail/svn-commits/2006-October/017486.html |
| 01:01.52 | *** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 01:06.55 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 01:07.14 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 01:18.21 | MRH2 | well gtg battery dying :) |
| 01:20.37 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:22.31 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 01:27.33 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:28.14 | *** join/#asterisk PoWeRKiLL (n=powerkil@80.178.76.83.adsl.012.net.il) |
| 01:28.18 | PoWeRKiLL | hi |
| 01:28.46 | PoWeRKiLL | someone using t38 passthrough on * 1.4 ? |
| 01:29.22 | PoWeRKiLL | I got [Oct 5 03:29:31] NOTICE[9423]: chan_sip.c:4947 process_sdp: No compatible codecs, not accepting this offer! |
| 01:31.32 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 01:32.37 | *** join/#asterisk fugitivo (n=ajf@190.48.188.206) |
| 01:32.40 | fugitivo | hello |
| 01:33.42 | fugitivo | what's a good network card compatible with digium hardware? |
| 01:34.10 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 01:34.43 | file | PoWeRKiLL: can I get access to the box? |
| 01:35.07 | kronic | how could I determine if a particular channel is associated with an agent in the dialplan? |
| 01:35.08 | file | I want to see if a mod will help |
| 01:35.38 | kronic | like a lookup method ? |
| 01:37.57 | PoWeRKiLL | file yes you can |
| 01:38.38 | fugitivo | anyone is experiencing a random sip freeze on asterisk? only way to fix it is restarting asterisk |
| 01:44.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:47.10 | *** join/#asterisk Yogik (n=Miranda@c-66-41-255-50.hsd1.mn.comcast.net) |
| 01:48.04 | Yogik | Hello , does anyone have any idea where I could get a documentation on multi-tenant setup , or some examples |
| 01:51.39 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 01:52.39 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 01:53.02 | wulfy814 | any polycom folks here? I want to factory reset the phone, I just choose format file system right? |
| 01:53.05 | rene1 | hello. i have a small question. using queues and mixmonitor i would like to change the default name of recordings. can this be done via -> Set(MONITOR_FILENAME=foo) a priority before entering the queue? |
| 01:53.10 | wulfy814 | that will take it back to original bootrom? |
| 01:54.38 | Yogik | with policom - just format |
| 01:54.43 | Yogik | polycom that is |
| 01:55.12 | wulfy814 | will I need to supply a bootrom or will it just go back to factory? |
| 01:55.19 | Yogik | you can get new bootrom from http://www.freedomphones.net/polycom/files/ |
| 01:55.29 | Yogik | no , no need for new bootrom |
| 01:55.32 | Yogik | but |
| 01:55.33 | Yogik | wait |
| 01:55.42 | Yogik | you will need new sip application |
| 01:55.50 | wulfy814 | I have sip.ld (1.67) |
| 01:55.55 | wulfy814 | it's IP600 |
| 01:56.05 | wulfy814 | I just don't have a bootrom on the server |
| 01:56.10 | Yogik | I see sip 2.0.2 |
| 01:56.13 | Yogik | wait |
| 01:56.15 | wulfy814 | FTP provisioning |
| 01:56.16 | Yogik | 2.0.1 |
| 01:56.21 | wulfy814 | I hear bad things about 2.0.1 |
| 01:56.22 | Yogik | ftp works |
| 01:56.36 | wulfy814 | I had phones rebooting with it after hanging up a SIP to SIP call |
| 01:56.39 | Yogik | well, when you format you will get your sip deleted |
| 01:56.49 | wulfy814 | right, so the 1.67 should work fine :-) |
| 01:57.02 | Yogik | yep I have IP500 with it |
| 01:57.08 | Yogik | works great with 3 lines |
| 01:57.18 | wulfy814 | I had originally mucked around with the webconfig |
| 01:57.28 | wulfy814 | and now some of those settings are overriding my good ftp configs |
| 01:57.35 | Yogik | right |
| 01:57.37 | Yogik | hehe |
| 01:57.59 | Yogik | put your put sip application on ftp |
| 01:58.03 | wulfy814 | it's there |
| 01:58.06 | Yogik | it will pull it buy default |
| 01:58.09 | Yogik | just format then |
| 01:58.13 | Yogik | bootrom will stay |
| 01:58.14 | wulfy814 | in process :-) |
| 01:58.37 | Yogik | do you know any good documentation for multi-tenant setup? |
| 01:58.40 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 02:00.43 | wulfy814 | unfortunately no |
| 02:01.08 | wulfy814 | ok another one, I have a phone that's showing the correct date, but the time is an hour off |
| 02:01.19 | gambolputty | rich |
| 02:01.23 | wulfy814 | they are all pulling from the same sip.cfg with the time offset correct |
| 02:01.31 | wulfy814 | three phones are ok, one wrong |
| 02:02.47 | Yogik | hmm |
| 02:02.57 | Yogik | same offset in config? |
| 02:03.01 | wulfy814 | yeah |
| 02:03.06 | wulfy814 | I'm going to reboot it for kicks |
| 02:03.21 | wulfy814 | I've never been in the webconfig on this phone |
| 02:03.34 | wulfy814 | and the menu on the phone itself doesn't have anything |
| 02:04.10 | Yogik | offset could be set in config , as far as I recall |
| 02:07.41 | *** join/#asterisk slobberknocker (n=slobberk@c-67-169-248-217.hsd1.ut.comcast.net) |
| 02:07.50 | slobberknocker | anyone here good with mysql queries? |
| 02:09.50 | Yogik | what do you need to do with mysql? |
| 02:10.40 | rene1 | Set(MONITOR_FILENAME=foo) worked for me |
| 02:11.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:11.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:12.50 | slobberknocker | i have my select statement all worked out except for one piece. I have a field named uniqueid and the values there are number like 1159974934.19308. the numbers before the . are not always unique. there could be up to 10 or so of the same number. but the number after the . is always unique. I am trying to list out the results only by the first portion of the number, the numbers before the "." how can i do this? |
| 02:13.02 | slobberknocker | here is my query: select distinct * from cdr where clid = 'efolks' and calldate between '2006-10-04 00:00:00' and '2006-10-04 23:59:59' and lastapp is not null; |
| 02:13.45 | slobberknocker | basically, i need to see how many unique calls i have |
| 02:14.22 | rene1 | sobberknocker |
| 02:14.24 | rene1 | do a grou by |
| 02:14.27 | rene1 | group by |
| 02:14.53 | Yogik | I see, well if it's the one filed I'd look into splitting to 2 fields into a temp table and then run a group by ( thanks rene1 ) query |
| 02:14.59 | rene1 | do a group by NUMBER in the where clause and in the SELECT clause do a SUM(duration) |
| 02:15.15 | rene1 | np |
| 02:15.35 | rene1 | a COUNT(*) |
| 02:15.39 | rene1 | is what you need |
| 02:15.45 | Yogik | rene - it's one filed |
| 02:15.49 | rene1 | filed? |
| 02:15.54 | Yogik | field |
| 02:15.56 | Yogik | sorry |
| 02:16.29 | rene1 | ahh |
| 02:16.39 | rene1 | does your database support regex? |
| 02:16.49 | rene1 | you could do it with a regex |
| 02:16.52 | slobberknocker | dunno... only started into mysql today :-D |
| 02:17.07 | rene1 | mysql 5 should have regex |
| 02:17.18 | rene1 | as for 4.x i am not sure but quite likely |
| 02:17.24 | slobberknocker | ok |
| 02:18.32 | slobberknocker | ok... so if my number is 1159974934.19320 is there not a cleaner way to just search by the 1159974934. portion of the value? |
| 02:18.40 | rene1 | the regex is not very complicated you basically want to do a substitution of a pattern like \.\d$ into "" |
| 02:18.41 | Yogik | hmm |
| 02:18.54 | rene1 | \.\d{3}$ |
| 02:18.56 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 02:19.02 | Yogik | let me take a look at docs |
| 02:19.06 | slobberknocker | what would that syntax look like? |
| 02:19.08 | rene1 | not a regex whiz mysself |
| 02:19.41 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 02:22.30 | Yogik | you may want to look here http://dev.mysql.com/doc/refman/4.1/en/string-comparison-functions.html |
| 02:23.17 | rene1 | slobberknocker why do you need to sort via the uniqueid |
| 02:24.51 | slobberknocker | well... that is what I thought would be the best way to do what I am looking for. I need to count all of the unique calls to a particular number. but in my data, each time a call is transferred and such there is a new line created... thus the second part of the uniqueid number. |
| 02:24.57 | slobberknocker | is there a better way to do what I need? |
| 02:25.21 | Yogik | there you go LEFT(str,len) - Returns the leftmost len characters from the string str. |
| 02:25.21 | Yogik | mysql> SELECT LEFT('foobarbar', 5); |
| 02:25.21 | Yogik | <PROTECTED> |
| 02:25.41 | *** join/#asterisk PoWeRKiLL (n=powerkil@85.64.221.118.dynamic.barak-online.net) |
| 02:26.07 | slobberknocker | let me try that... thanks |
| 02:26.23 | Yogik | so do a group by ( Left("fieldname", 10 ) |
| 02:26.36 | *** join/#asterisk alexmontoanelli (n=alexm@alexmm.unetvale.com.br) |
| 02:28.45 | rene1 | cool |
| 02:29.22 | Yogik | but I suppose it will be very slow :) |
| 02:31.35 | *** join/#asterisk mat2 (n=mat@c-71-198-139-13.hsd1.ca.comcast.net) |
| 02:31.50 | slobberknocker | double chek my work: select * from cdr where clid = 'efolks' and calldate between '2006-10-04 00:00:00' and '2006-10-04 23:59:59' group by (Left('uniqueid', 10)); |
| 02:31.54 | slobberknocker | i only got one row back... there should be 50 or so |
| 02:32.03 | mat2 | hello everyone |
| 02:34.39 | mat2 | i just upgraded to the latest version of asterisk, and Im getting an error with my music on hold. i get the following error: |
| 02:34.41 | mat2 | <PROTECTED> |
| 02:34.41 | *** join/#asterisk ingenius (n=syntax@93-13-235-201.fibertel.com.ar) |
| 02:35.01 | mat2 | these files worked previously, and the files are definitely there |
| 02:35.27 | Yogik | <PROTECTED> |
| 02:35.27 | mat2 | has anyone else had problems? |
| 02:35.46 | Yogik | permissions? |
| 02:36.02 | slobberknocker | yeah... permissions... had that issue before |
| 02:36.35 | Yogik | chown -Rv asterisk:asterisk /var/lib/asterisk |
| 02:36.49 | mat2 | heres my current perms: |
| 02:36.50 | mat2 | -rw-r----- 1 root asterisk 2865614 May 15 16:02 Eric Clapton - 32-20 Blues.mp3 |
| 02:36.50 | mat2 | -rw-r----- 1 root asterisk 3446996 May 15 16:02 Eric Clapton - Come On In MyKitchen.mp3 |
| 02:36.50 | mat2 | -rw-r----- 1 root asterisk 3711564 May 15 16:02 Eric Clapton - Hellhound On My Trail.mp3 |
| 02:37.15 | Yogik | what about directory permissions? |
| 02:37.26 | *** join/#asterisk alexmontoanelli (n=jircii@alexmm.unetvale.com.br) |
| 02:37.43 | mat2 | drwxr-x--- 2 asterisk asterisk 4096 May 15 16:02 matjones |
| 02:37.54 | mat2 | drwxr-x--- 5 asterisk asterisk 4096 May 15 16:06 mohmp3 |
| 02:38.17 | Yogik | and your asterisk is running as? |
| 02:38.22 | mat2 | asterisk |
| 02:38.25 | Yogik | ps -aux | grep asterisk |
| 02:39.30 | mat2 | actually. might be root |
| 02:39.30 | mat2 | root 17987 0.0 1.8 15048 7272 ? S 18:07 0:00 asterisk -cvvvvvvvvvvvvv |
| 02:39.58 | Yogik | then it's not permission problems, you are runnign as root |
| 02:41.18 | Yogik | you using mpg123 as application for moh? |
| 02:42.45 | mat2 | im just using the default method I believe |
| 02:43.17 | mat2 | I found this article, but not sure if its the same thing. I dont have a capi.conf file |
| 02:43.18 | mat2 | http://www.mail-archive.com/openpbx-dev@openpbx.org/msg00330.html |
| 02:43.36 | mat2 | it used to work fine pre-upgrade |
| 02:44.20 | slobberknocker | select uniqueid from cdr where clid = 'efolks' and calldate between '2006-10-04 00:00:00' and '2006-10-04 23:59:59' group by (Left('uniqueid', 10)); sill only returns one row |
| 02:44.23 | slobberknocker | still only... |
| 02:45.11 | *** join/#asterisk jtoy_ (n=jtoy@c-24-63-128-84.hsd1.ma.comcast.net) |
| 02:45.40 | jtoy_ | what is the best open source GUI that can be installed on top of a regular asterisk? |
| 02:45.56 | jtoy_ | there seem to be so many options and most of them arent open source |
| 02:46.43 | rene1 | jtoy_: why do you need a gui? for your end users? |
| 02:47.05 | orlock | jtoy_: freepbx |
| 02:47.08 | rene1 | see systems like avaya or alcatel no not have guis |
| 02:47.27 | rene1 | thats why there exist maintainance contracts |
| 02:47.35 | jtoy_ | rene1: the person who will be admining the server in the long run doesnt know too much of asterisk so i wanted to start them on something easier in the beginning |
| 02:47.56 | rene1 | teach him how to edit sip.conf |
| 02:48.00 | rene1 | that is not very difficult |
| 02:49.08 | mat2 | yes. i started with a gui. but quickly learned it was a waste of time, and should edit text files to get asterisk to do exactly what you want |
| 02:49.57 | jtoy_ | also, is there gui for the desktop that users can use to dial from and see a caller id when people call, some gui that works directly with the phone associated with that computer? |
| 02:50.21 | jtoy_ | that is the most important feature that the sales teams for when we cdhange phone systems |
| 02:50.47 | rene1 | jtoy: i have seen those callerid panels for windows |
| 02:50.49 | rene1 | they are cool |
| 02:51.00 | rene1 | take a look at the voip-info wiki |
| 02:51.07 | rene1 | under graphical interfaces |
| 02:51.24 | rene1 | and well then there is outlook integration for asterisk |
| 02:51.31 | rene1 | that can be very useful too |
| 02:51.35 | jtoy_ | there is so much info and most arent open source, icant tell |
| 02:51.47 | jtoy_ | yeah, our sales guys use outlook |
| 02:51.48 | rene1 | some are |
| 02:52.03 | jtoy_ | im afraid that if the new person admins asterisk, they might change one line and mess the whole system up, |
| 02:52.14 | jtoy_ | that has happened with me many times in the past |
| 02:52.48 | jtoy_ | I always felt the asterisk conf files are crazier then sendmail, which is scary!!!!! |
| 02:53.17 | jtoy_ | I know the Flash Operator Panel exists, but thats not exactly the same |
| 02:53.25 | rene1 | no sendmail is scarier |
| 02:53.34 | rene1 | to me at least |
| 02:53.36 | rene1 | heh |
| 02:53.52 | rene1 | use freepbx then |
| 02:54.56 | jtoy_ | from reading about freepbx, is it supposed to go on top of any asterisk, or does it come with a version and can only be used with that verison, i cant tell too well. |
| 02:55.17 | Yogik | you can install it on latest build of asterisk |
| 02:55.37 | Bobcat991966 | Question. I was reading your converstaion erlier about permission and took a look at mine to see what user asterisk was running under. Is it normal to have safe_asterisk running? |
| 02:55.52 | Bobcat991966 | root 1598 0.0 0.1 4380 660 ? S 22:58 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 02:56.35 | Bobcat991966 | what exaclty is safe asterisk |
| 02:58.30 | slobberknocker | ok... i finally got it to work: select distinct left(uniqueid,10) from cdr where clid = 'efolks' and calldate between '2006-10-03 00:00:00' and '2006-10-03 23:59:59'; |
| 03:05.18 | *** join/#asterisk alexmontoanelli (n=alexmont@alexmm.unetvale.com.br) |
| 03:12.17 | *** join/#asterisk Schulich (n=Jazba@165.154.37.76) |
| 03:29.30 | *** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir) |
| 03:31.17 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 03:32.11 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:35.59 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 03:36.01 | JunK-Y | yo yo yo |
| 03:36.38 | file | JunK-Y: you're hurting me! |
| 03:37.07 | JunK-Y | no, bb is hurting me! |
| 03:37.14 | file | :D |
| 03:37.16 | file | yo yo Junky |
| 03:40.48 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.14.32.revip2.asianet.co.th) |
| 03:41.08 | HaMYaI | is there a particular wave file format for asterisk? |
| 03:41.50 | HaMYaI | I use a file in an unsigned 16bit mono, 8000Hz and the volume is very low |
| 03:42.28 | *** join/#asterisk Joel1978 (n=Joel1978@12-226-85-195.client.mchsi.com) |
| 03:42.42 | HaMYaI | compared to gsm at the same freq |
| 03:44.29 | *** join/#asterisk Blackthorn (n=blacktho@w-l4.smyth.net) |
| 03:45.19 | Blackthorn | off-topic question but was thinking somone could directly me. Is there a linix package/service that allows you to bridge ether/ether and packetshape? |
| 03:45.30 | Blackthorn | er direct |
| 03:50.14 | *** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 03:51.08 | *** join/#asterisk bmg505 (n=leon@c1-130-15.rndf.isadsl.co.za) |
| 03:52.54 | *** join/#asterisk Joel1978 (n=Joel1978@12-226-85-195.client.mchsi.com) |
| 03:58.25 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 03:58.29 | sudhir492 | Hi All |
| 03:58.34 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:59.05 | sudhir492 | Anyone here who has successfully implemented Paging on Polycom phones? |
| 04:00.17 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 04:00.21 | tessier_ | Hello all! |
| 04:00.47 | tessier_ | Anyone know if you can detect dropped RTP packets with tethereal? |
| 04:00.59 | tessier_ | I have a remote office claiming poor audio quality problems but I cannot find the real source of the issue. |
| 04:01.05 | orlock | hmm |
| 04:01.13 | orlock | tessier_: wireshark will do it |
| 04:01.25 | tessier_ | I would like to be able to quantify when and how it happens without being there |
| 04:01.26 | orlock | you need to dump the rtp stream and then load it up |
| 04:01.30 | tessier_ | I do have access to their firewall |
| 04:01.30 | orlock | yeah |
| 04:01.33 | orlock | tcpdump at remote site |
| 04:01.43 | orlock | scp data to somewhere you have wiresharl |
| 04:01.46 | tessier_ | wireshark == tethereal? I've been confused about that |
| 04:01.53 | orlock | ethereal |
| 04:02.00 | orlock | tethereal is the console version |
| 04:02.02 | tessier_ | I've always used ethereal and now this wireshark thing has popped up |
| 04:02.05 | orlock | yeah |
| 04:02.08 | orlock | same thing, name changed |
| 04:02.12 | tessier_ | ah, ok |
| 04:02.15 | orlock | copywrite issues iirc |
| 04:02.24 | tessier_ | It'll be the ruin of us all. |
| 04:02.31 | orlock | it has a pretty nice rtp diagnostings thing in it |
| 04:02.40 | tessier_ | So once I have the stream dumped and loaded up into ethereal how will I be able to tell if packets were dropped? |
| 04:02.51 | orlock | select thestream, and it gives you jitter and delay, plus any out of order packets |
| 04:03.02 | tessier_ | Neato. I'll do that. |
| 04:03.35 | tessier_ | Since RTP can be on any port is there any way I can tell it to only grab the SIP/RTP? |
| 04:04.00 | tessier_ | If it's really smart it can look at the SIP packets and discern what other traffic will be RTP |
| 04:04.02 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.136.106) |
| 04:04.05 | orlock | Statistics -> RTP -> |
| 04:04.09 | AJaymn | When someone calls in from my Sip Provider Im getting there Caller ID NAME, but my SIP # ... when I do sip debug its showing there # is comming across.. Any idea how to get it to show there # and not my trunks? |
| 04:04.23 | orlock | tessier_: yeah, it looks at the headers |
| 04:04.26 | tessier_ | nice |
| 04:04.41 | orlock | and it only does rtp, not sip stuff |
| 04:04.54 | orlock | well, it knows sip, etc, but the stream decoder only boters with rtp |
| 04:05.00 | orlock | you ca even dump the payload to wav |
| 04:05.06 | orlock | and play it back |
| 04:06.33 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.136.106) |
| 04:06.38 | Joel1978 | i'm having trouble choosing a linux distro....anyone willing to give their two cents as to which one i should choose for a small home office setup? |
| 04:07.06 | linuxbangalore | hi can ask about festival speech synthesis over here.. |
| 04:07.14 | orlock | Joel1978: centos |
| 04:07.46 | Joel1978 | ok that's what i normally use...so i'll just stick with that |
| 04:07.51 | gambolputty | centos |
| 04:08.47 | Joel1978 | thanks! |
| 04:09.29 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:09.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:10.38 | tessier_ | centos for servers and fedora for desktops IMHO but it's a religious issue |
| 04:10.45 | Joel1978 | heh |
| 04:10.51 | tessier_ | But I use them both in big production setups that make my company millions so it can definitely work |
| 04:11.21 | *** join/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 04:14.52 | sudhir492 | Anyone using Polycom phones here who has successfully implemented paging on them |
| 04:15.41 | sudhir492 | Joes1978: I have been using RH9 and FC3, even FC4. They all work well. |
| 04:16.18 | Joel1978 | cool |
| 04:16.30 | sudhir492 | I have been using FC3 on quite a few heavyweight servers, and they have been going strong |
| 04:17.20 | Joel1978 | is there a guide online as to which packages to install? |
| 04:17.26 | Joel1978 | for centos |
| 04:18.13 | Joel1978 | i have 4.4 burnt to CD already so i'll just use that |
| 04:20.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:24.17 | [hC] | unreal |
| 04:24.20 | [hC] | fonality bought trixbox |
| 04:24.31 | Joel1978 | sad isn't it |
| 04:29.11 | Joel1978 | that move has prompted me to buckle down and do a fresh install on a minimal linux install so i can really learn it all |
| 04:29.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:30.42 | *** join/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 04:36.58 | xai | What about a dual homed * box, that is hanging off DMZ network (IAX NAT to outside), and has a LAN IP for direct phone connection? Is that sensible? It removes the SIP-NAT problem.. |
| 04:37.57 | Joel1978 | when i setup my trixbox for testing i put it on the DMZ side and had someone in another city login with their softphone....worked fine |
| 04:40.29 | xai | Im mostly worried about Lan-> DMZ nat issues.. |
| 04:40.52 | xai | running SIP over the intenet is abhorent.. |
| 04:41.00 | Joel1978 | as long as your * box as an internal ip you'll be fine |
| 04:41.30 | Joel1978 | the router should manage that for you... |
| 04:41.32 | *** join/#asterisk AJay-mn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 04:41.37 | xai | Joel1978: so I'd have to dual home the * box for that.. |
| 04:41.51 | Joel1978 | what sort of router are you using? |
| 04:42.03 | xai | Or static routing.. ? |
| 04:42.06 | xai | m0n0wall .. |
| 04:42.29 | xai | I can do all sorts of NAT or whatever. i thought SIP didn't like nat transversal. |
| 04:42.32 | Joel1978 | you could do nat with internal static routes for standard ports |
| 04:43.10 | [hC] | id love to chime in and help but you guys are throwing the wrong terms around by the looks of it |
| 04:43.18 | [hC] | by static routing, do you mean port forwarding? |
| 04:43.44 | xai | no. i mean static routing between 2 private IP spaces. |
| 04:44.03 | xai | ie. LAN -> dmz |
| 04:44.54 | [hC] | what exactly do you mean by dmz? a publically routable ip? |
| 04:45.14 | Joel1978 | demilitarized zone |
| 04:45.31 | xai | dmz is usually on private space. |
| 04:45.42 | Joel1978 | allowing all incoming traffic to be routed to an internal IP |
| 04:46.03 | [hC] | i know what dmz stands for, but it didnt make sense how you were talking about doing it, so i was curious what you were trying to accomplish |
| 04:46.43 | xai | Joel1978: i think i can get better performance off my dual homed scenario, because the Lan-> * doesnt go through the firewall, and there is no nat, so that reduces latency too. |
| 04:46.57 | sbingner | dmz is something that's protected from the internet but your network is protected from it |
| 04:47.51 | Joel1978 | if you're concerned about security, at nat port forwarding |
| 04:48.01 | Joel1978 | otherise just go with the dmz setup |
| 04:48.12 | Joel1978 | at = setup |
| 04:49.43 | [hC] | so... you are talking about putting one nic in a DMZ on a completely isolated subnet, and utilizing a routers 'DMZ' functionality in that it forwards all ports to it, and having your phones on the LAN on another subnet and physical nic? |
| 04:50.12 | tessier_ | Holy shit |
| 04:50.15 | Joel1978 | ha |
| 04:50.22 | Joel1978 | that sounds like a mess to me |
| 04:50.23 | tessier_ | ethereal/wireshark has improved a whole hell of a lot |
| 04:50.34 | [hC] | tessier: especially for SIP/RTP tracing.. |
| 04:50.37 | tessier_ | It's been a year or two since I used it to look at SIP/RTP |
| 04:50.47 | tessier_ | [hC]: Exactly |
| 04:50.59 | [hC] | Joel1978: thats why im asking, is that what you're describing? |
| 04:51.08 | tessier_ | Draws a graph of the whole sip conversation, finds associated RTP, does jitter/loss analysis, and recombines the streams both ways in to a raw file! |
| 04:51.42 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 04:52.12 | [hC] | Joel1978: i do installs similarly to tht, but not exactly that way. but unless you need to connect phones to your * box from outside the LAN, dont even bother. routers these days will handle things just fine. |
| 04:52.23 | [hC] | plus, you're not going to have a nat/sip issue internally, only if you need to register outbound |
| 04:52.31 | [hC] | and in that circumstance, just use IAX. |
| 04:52.41 | Joel1978 | right |
| 04:56.24 | xai | Well, it's only going to talk IAX on the intenet side.. |
| 05:01.45 | *** join/#asterisk DoDaT69 (i=Tracer@66.240.110.115) |
| 05:02.41 | xai | Offices connect to each other's * (each on dmz) via iax. The phones live in LAN and talk to DMZ, but that seems like a waste of NAT and routing power. |
| 05:03.13 | [hC] | err... wtf.. i just sent a polycom a notify to reboot, it grabbed all its configs, never re-registered and now i cant ping it.. |
| 05:03.17 | [hC] | thats pretty damn strange. |
| 05:05.27 | xai | http://pastebin.com/800510 |
| 05:06.40 | DoDaT69 | how in the hell do I configure an analog sangoma card? |
| 05:06.44 | DoDaT69 | I am having hella problems |
| 05:08.04 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:08.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:12.49 | DoDaT69 | does anyone here have any experience with these sangoma analog cards? |
| 05:14.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 05:26.56 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 05:27.17 | aadilismail | how to change the SIP Port of asterisk? |
| 05:28.47 | Mavvie | aadilismail: if it can be done it can be done in the /etc/asterisk/sip.conf file |
| 05:28.49 | Mavvie | search for 5060 |
| 05:29.08 | aadilismail | alright |
| 05:30.25 | Joel1978 | under general look for port=5060 |
| 05:30.29 | Joel1978 | in sip.conf |
| 05:31.34 | aadilismail | it's done....bindport=5060 |
| 05:31.35 | aadilismail | thanks |
| 05:32.26 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
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| 06:02.10 | FuriousGeorge | is anyone using sipphone? |
| 06:02.34 | Joel1978 | i do |
| 06:03.37 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 06:04.19 | *** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 06:04.25 | AJaymn | Anyone using Broadvoice? |
| 06:04.40 | x86 | AJaymn: heya :) |
| 06:04.54 | AJaymn | hey ;) |
| 06:05.39 | FuriousGeorge | Joel1978: hows the quality |
| 06:05.47 | FuriousGeorge | thier rate is 1 cent a minute |
| 06:05.52 | FuriousGeorge | thats rediculously cheap |
| 06:06.36 | Joel1978 | yeah it's ok i guess....i haven't used it much |
| 06:06.44 | Joel1978 | i use sipplan.com mainly |
| 06:07.19 | Chris-NB | hi |
| 06:07.31 | Chris-NB | anyone played around with dns enum entires? |
| 06:08.01 | FuriousGeorge | Joel1978: yeah, i need it for business purposes |
| 06:08.14 | Chris-NB | can someone explain to me, how a range of extentions from 20-30 has to be entered? |
| 06:08.51 | AJaymn | chris-nb Very carefully ;) |
| 06:09.15 | Chris-NB | AJaymn, do I have to enter each extention manually? |
| 06:09.17 | Joel1978 | FG: well they're both prepaid services, so if you don't like, just switch to another |
| 06:09.32 | Chris-NB | AJaymn, or is it possible to use regex? |
| 06:09.45 | AJaymn | chris-nb I was being a smart ass.. I have no idea ;) I enter all mine manually |
| 06:10.00 | Joel1978 | FG: sipplan seems to be very responsive whereas sipplan takes a few extra rings to complete the call |
| 06:10.43 | Chris-NB | AJaymn, ok : ) |
| 06:11.05 | AJaymn | its late and ive been on Mountain Dew for 8 hours ;) |
| 06:13.44 | Joel1978 | has anyone tested out the new chan_skype? |
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| 06:23.53 | rene1 | woooo |
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| 06:37.47 | DoDaT69 | my zaptel channels arent showing up in asterisk, but ARE showing when I do ztcfg -vvvv |
| 06:38.07 | DoDaT69 | how do I get em to come up in asterisk?!? |
| 06:40.49 | *** join/#asterisk Rhizome (n=Rhizome@host-81-191-151-89.bluecom.no) |
| 06:40.53 | sbingner | configure them in zapata.conf? |
| 06:41.30 | DoDaT69 | yup |
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| 06:51.56 | *** join/#asterisk SoftIce (n=awk@vc-196-207-45-253.3g.vodacom.co.za) |
| 06:52.15 | SoftIce | hello, how can I configure my firewall to allow RTP to transfere over it |
| 06:52.26 | SoftIce | It says it doesn't use TCP or UDP |
| 06:52.38 | x86 | RTP goes over UDP |
| 06:52.49 | SoftIce | RTP does not have a standard TCP or UDP port that it communicates on |
| 06:52.55 | SoftIce | ahh |
| 06:53.00 | x86 | right, it uses a range of ports |
| 06:53.02 | SoftIce | sorry, I must have mis understood what they where saying. |
| 06:53.08 | x86 | as defined in rtp.conf in /etc/asterisk |
| 06:53.16 | SoftIce | so I just need to open 10000:20000 |
| 06:53.26 | x86 | whatever port range you have in rtp.conf, yes |
| 06:53.27 | SoftIce | thats a big range to open ;) |
| 06:53.31 | x86 | well you can change it |
| 06:53.42 | x86 | you need one port per concurrent channel in use |
| 06:53.49 | SoftIce | but i'm not sure how many ports it will need to use |
| 06:53.58 | x86 | so you can bring that number of ports down considerably |
| 06:54.23 | x86 | well you can make the range be about 100 ports or so |
| 06:54.39 | x86 | or even 1000 if you really wanted to |
| 06:55.47 | SoftIce | ye, kewl i'll need about 1000 for starters |
| 06:55.52 | SoftIce | thanks x86 |
| 06:56.32 | SoftIce | x86: do yoou have your SIP connections using UDP or TCP? |
| 06:56.36 | x86 | you know a single Asterisk box wont be able to handle much more than 100 calls anyway right? :) |
| 06:56.40 | SoftIce | I know TCP requires more work |
| 06:57.07 | x86 | do it over UDP |
| 06:57.48 | SoftIce | right, what about the asterisk daemon port 2000 is that needed to be viewed at all publicly if its going to be a peer ? |
| 06:57.52 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:57.59 | SoftIce | no other peer, etc will need to talk to the daemon directly? |
| 06:58.22 | x86 | port 2000... not sure what that is... perhaps it's the manager port |
| 06:58.43 | SoftIce | ye, the manager |
| 06:58.51 | x86 | in which case you wont need unless you're using applications that remotely access the manager interface |
| 06:59.03 | SoftIce | right, kewl. |
| 06:59.17 | SoftIce | 1 last thing :) |
| 06:59.28 | SoftIce | changing the rtp.conf configure on a range, wont need to be changed anywhere else? |
| 06:59.35 | x86 | nope |
| 06:59.38 | SoftIce | so if I change the range I wont need to configure anything else to pass on that range? |
| 06:59.41 | SoftIce | ah, kewl. |
| 06:59.44 | x86 | you will have to restart asterisk though |
| 06:59.52 | x86 | not sure if a simple reload will do it or not |
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| 07:10.44 | *** part/#asterisk Emrah (n=Me@adslgva0491.worldcom.ch) |
| 07:14.27 | SoftIce | pfft! as soon as I start the firewall |
| 07:14.29 | SoftIce | I get no sound |
| 07:14.37 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:14.44 | SoftIce | even though the ports are open under UDP |
| 07:15.17 | Rhizome | Anyone have any idea why the mysql module will fail constantly with "res_config_mysql.so: undefined symbol: __pure_virtual ? |
| 07:15.48 | SoftIce | it means mysql wanst installed properly |
| 07:16.15 | SoftIce | did you install the addons pack? |
| 07:16.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:16.40 | Rhizome | ofcourse, thats when it starts crashing, i'll try reinstalling mysql ;) |
| 07:17.22 | SoftIce | it starts crashing when you installed the addons? |
| 07:17.28 | SoftIce | you get res_config_mysql ? |
| 07:17.38 | SoftIce | .so undifed bla bla bla? |
| 07:17.39 | Rhizome | after I install the addon and try to start asterisk |
| 07:18.09 | SoftIce | do you have the res_mysql.conf file in place? |
| 07:19.14 | *** join/#asterisk MacWinner (n=chatman@bb219-74-38-112.singnet.com.sg) |
| 07:19.16 | SoftIce | and are you SURE you using the latest version of asterisk ? |
| 07:19.45 | Rhizome | yea just copied over res_mysql.conf |
| 07:20.11 | Rhizome | Just wondering where __pure_virtual comes from. |
| 07:20.24 | MacWinner | hi all, i wanted to setup a PBX for our new office with about 30-40 people.. could you suggest a tutorial or simple site to do this? all we need right now is a VoIP calls through a VoIP trunk, and voicemail.. then I'll add IVR etc later |
| 07:20.42 | SoftIce | ye, i'm not sure |
| 07:20.56 | MacWinner | i remember there is a boot CD for asterisk that basically does everything for you.. what was that called? |
| 07:21.18 | MacWinner | oh.. asterisk@home |
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| 07:25.42 | Joel1978 | trixbox is actually the name for it now |
| 07:25.53 | Joel1978 | http://trixbox.org |
| 07:26.48 | MacWinner | cool, thanks! |
| 07:26.57 | Joel1978 | np |
| 07:27.03 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:27.25 | SoftIce | blah, there I am on my day off, SICK and my boss phones me to open up some rules! |
| 07:27.27 | MacWinner | is there some sort of physical wifi phones (like a skype phone), that you can use for asterisk? |
| 07:27.29 | SoftIce | i think that is pretty blind ;( |
| 07:27.37 | orlock | SoftIce: always happens to me |
| 07:27.42 | orlock | SoftIce: holday, sick.. anything |
| 07:27.46 | SoftIce | hahah |
| 07:27.51 | SoftIce | you put a smile on my face |
| 07:27.56 | orlock | i cant remember the last time i had time off and didnt get a call! |
| 07:28.02 | MacWinner | for example, i want to setup a small linux box with some wifi phones in the office, and then give them these wifi handsets |
| 07:28.09 | Joel1978 | mac: linksys is one company that makes them |
| 07:28.11 | MacWinner | that connnect to the asterisk pbx |
| 07:28.13 | Joel1978 | they're expenisve tho |
| 07:28.16 | orlock | 11:30pm sititng in a farm surrounded by cows next to a fire, one bar of mobile reception.. |
| 07:28.26 | orlock | "Hi, i am a technician doing some work at so-and-so" |
| 07:28.32 | MacWinner | Joel1978: which one from linksys? |
| 07:28.45 | Joel1978 | http://www.voip-info.org/wiki/view/Wireless+VOIP |
| 07:28.54 | MacWinner | i wouldn't mind buying it if it's cheaper than buying the server |
| 07:29.00 | Joel1978 | not sure off the top of my head but that link will give u a starting point |
| 07:29.14 | SoftIce | orlock: iesh, thats hectic man! ;) |
| 07:29.39 | MacWinner | i'm currently in singapore, and they have this cool 3G-to-wifi linksys device so you can have highspeed anywhere and use wifi to access it |
| 07:29.42 | Joel1978 | you're looking at the low $100 - upper $200 |
| 07:30.10 | MacWinner | oh.. i'm willing to go up to $1000 if i get some nice configurability |
| 07:30.23 | Joel1978 | heh |
| 07:30.28 | Joel1978 | wish i had your budget |
| 07:30.54 | MacWinner | company budget :) |
| 07:31.38 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 07:32.16 | MacWinner | was the linksys device you were thinking of for use in an office |
| 07:32.22 | jmls | OT - is there a room history, so that when I log in again I get the last xxx messages ? I am often swapping locations and don't get all the messages during the times I am offline |
| 07:32.22 | MacWinner | ? like with incoming and everything |
| 07:33.32 | Joel1978 | it's a sip compatible phone....as for it's acceptible use as an office phone i'm not so sure. that's something you'll have to review =) |
| 07:35.32 | MacWinner | maybe it's just better to get everyone the skype phones and connect them to our wifi |
| 07:35.47 | MacWinner | everyone will have their own phone number |
| 07:36.08 | orlock | MacWinner: get some standard cordless decta phones and an ATA |
| 07:36.20 | Joel1978 | asterisk/trixbox has so much more functionality |
| 07:36.37 | Joel1978 | and you can probably save the company money in the end on line charges |
| 07:39.05 | *** join/#asterisk muther (n=jubei@147.27.27.27) |
| 07:39.52 | kaldemar | hello |
| 07:40.31 | SoftIce | dam I can't believe how big skype has become |
| 07:40.32 | SoftIce | ;p |
| 07:40.35 | kaldemar | would anyone have done emacs syntax coloring for the extensions.conf syntax? |
| 07:41.00 | SoftIce | why would you want to do that? :P |
| 07:41.08 | SoftIce | and who uses emacs :) |
| 07:41.19 | Joel1978 | lol |
| 07:41.23 | x86 | EmacsOS is a PITA |
| 07:41.30 | x86 | vim++ |
| 07:41.33 | Joel1978 | cause it has color coded markup which is nice |
| 07:41.47 | x86 | so does vim, and joe, and jed |
| 07:41.52 | Joel1978 | didn't know that |
| 07:42.04 | x86 | and so many other sane editors |
| 07:42.07 | SoftIce | exactly |
| 07:42.08 | Joel1978 | they should add that to nano |
| 07:42.14 | x86 | nano is lame |
| 07:42.16 | kaldemar | SoftIce: if you use a bit more complex structures with lots of variables and arithmetic operations, it would make writing macros much easier. |
| 07:42.16 | Joel1978 | lol |
| 07:42.17 | x86 | as is pico |
| 07:42.20 | SoftIce | nano ;) |
| 07:42.21 | Joel1978 | why is it lame? |
| 07:42.23 | SoftIce | I like pico too |
| 07:42.40 | x86 | nano == a true open source version of pico |
| 07:42.42 | SoftIce | pico == nano ? |
| 07:42.45 | x86 | right |
| 07:42.52 | Joel1978 | aye |
| 07:42.54 | x86 | but pico is not truely "free" |
| 07:43.07 | MacWinner | wow.. i'm seriously looking at just using skype in for each of our people.. $38 per year per person.. and skypeOUT will just be metered... and skype-to-skype between offices is free |
| 07:43.08 | x86 | it has a proprietary license from the university of washington |
| 07:43.21 | Joel1978 | licensing is such a headache |
| 07:43.23 | kaldemar | so, no. :) |
| 07:43.36 | SoftIce | MacWinner: tsk tsk tsk |
| 07:44.14 | MacWinner | hehe.. whatever gets the job done man :) |
| 07:44.35 | Joel1978 | well if you wish to be limited...sky is the way =) |
| 07:44.55 | SoftIce | well I know a number of people who have setup their own asterisk box |
| 07:45.02 | SoftIce | just for their company |
| 07:45.07 | SoftIce | why not do it yourself? |
| 07:45.27 | kaldemar | http://archives.free.net.ph/message/20051021.154841.6a6cbd8c.en.html |
| 07:45.36 | kaldemar | in case someone else is interested... |
| 07:49.50 | *** join/#asterisk _omer (n=_omer@202.166.161.23) |
| 07:49.53 | _omer | hi |
| 07:50.39 | _omer | anyone who could help for Cisco AS5350 ? |
| 07:50.49 | x86 | #cisco |
| 07:51.15 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 07:51.27 | _omer | thanks |
| 07:51.32 | *** part/#asterisk _omer (n=_omer@202.166.161.23) |
| 07:51.44 | Stephnie | hello everybody |
| 07:53.11 | Stephnie | my SIP Carrier has given me 2 different IP Addresses (one for signalling and another for RTP) ..so how can I use these two IPs in SIP Peer? |
| 07:53.47 | x86 | that's bizarre ;) |
| 07:54.49 | SoftIce | forwarding |
| 07:55.08 | Stephnie | yep...confusing |
| 07:55.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:55.21 | Stephnie | :-/ |
| 07:55.58 | Stephnie | x86: is it possible in asterisk??? |
| 07:56.14 | edwar64896 | Stephanie: the signalling should indicate the RTP path - ip address and port |
| 07:56.25 | edwar64896 | if it is different, then surely SIP should handle this? |
| 07:57.07 | Stephnie | yes....their signalling should handle the RTP path....but what to do now ? |
| 07:58.13 | Stephnie | I tried to send calls at both of IP Address but no response...as my carrier says...RTP IP doesnt reply to INVITE MMESSAGE :S |
| 07:58.17 | SoftIce | why not setup a proxy to forward that IP |
| 07:58.45 | Stephnie | does asterisk can do proxy ? |
| 07:59.31 | hank | i have two hfc cards, one in te and one in nt mode. how would i configure the signalling method in zapata.conf? |
| 08:00.19 | SoftIce | Stephnie: you have to implement some sort of pass through, eg; a DMZ pass through |
| 08:00.35 | muther | guys I'm having trouble registering a sip softphone to my asterisk server. Should I be able to telnet to port 5060 if the FW is open? |
| 08:01.15 | Stephnie | SoftIce: any URL to get more detail? |
| 08:01.26 | SoftIce | muther: no you can't telnet the port |
| 08:02.05 | SoftIce | Stephnie: you have to research IP binding |
| 08:02.14 | SoftIce | NAT could be an option |
| 08:03.42 | *** join/#asterisk xnon (n=xnon@200.8.87.1) |
| 08:04.20 | SoftIce | Stephnie: look into using Squid.. |
| 08:04.33 | SoftIce | but I know routing can also be an option |
| 08:04.46 | SoftIce | as you can pass the route from your 1 IP to another and visa versa |
| 08:07.18 | Stephnie | are you sure thats for me what you are typing here :) |
| 08:07.43 | Stephnie | my carrier wants me to send INVITE at different IP and RTP at different IP |
| 08:07.46 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.136.106) |
| 08:08.01 | linuxbangalore | hi.. may I what is CLI mode in asterisk |
| 08:08.06 | muther | I can't seem to be able to register to my asterisk from a softphone (Ekiga). Here's my sip.conf http://pastebin.ca/191713 |
| 08:08.07 | linuxbangalore | and how to start asterisk in that mode |
| 08:08.25 | Rhizome | linuxbangalore: just start asterisk and then type asterisk -r |
| 08:08.36 | *** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it) |
| 08:08.44 | SoftIce | Stephnie: well what i'm trying to say is I have no idea how to do it with Asterisk |
| 08:08.52 | linuxbangalore | ok |
| 08:09.21 | SoftIce | so what i'm saying is you can pass a port range eg: UDP traffic from your asterisk server IP to your second IP you have that will route the traffic to them |
| 08:09.22 | Stephnie | :) |
| 08:09.26 | SoftIce | i'm giving you an option. |
| 08:09.33 | *** part/#asterisk RestLessGemini (n=rLg@202.61.49.248) |
| 08:10.14 | Stephnie | u mean only RTP? |
| 08:10.39 | SoftIce | yes |
| 08:10.45 | SoftIce | well what ever the RTP range is |
| 08:10.56 | Stephnie | and signalling ? |
| 08:11.01 | SoftIce | you can just pass that range, route it through the second IP |
| 08:11.18 | SoftIce | Stephnie; as normall |
| 08:11.41 | linuxbangalore | muther: I think you are missing the context value in [jubei] |
| 08:12.19 | linuxbangalore | and set the dtmfmode=rfc2833 and qualify=yes |
| 08:12.31 | muther | but i've put it general, doesn't that count for all peers?:) |
| 08:13.21 | Stephnie | how to send only RTP to my second IP ? |
| 08:13.45 | SoftIce | Stephnie: well what range does RTP use? |
| 08:14.07 | SoftIce | route all info from this IP on this port to the second IP |
| 08:14.23 | Stephnie | I see! |
| 08:14.42 | muther | linuxbangalore, I tried the changes you proposed but i still get registration failed :/ |
| 08:14.42 | Stephnie | need a port forwarding as well |
| 08:14.48 | SoftIce | eg: 10.0.0.1 = 1234:1334 -> 10.0.0.2 -> 192.10.0.1 |
| 08:15.10 | SoftIce | Stephnie: yes, you must play with routing and port forwarding. |
| 08:15.19 | SoftIce | if you using fbsd, try use somethhing like pf |
| 08:15.25 | Stephnie | but RTP could use any non-standard port.. |
| 08:15.43 | SoftIce | Stephnie: no it uses a range in /etc/asterisk/rtp.conf |
| 08:15.47 | muther | linuxbangalore, and sip show peers shows jubei as unreachagle |
| 08:15.48 | SoftIce | check the range and forward just that range. |
| 08:15.50 | muther | unreachable* |
| 08:16.02 | Stephnie | let me check plz....brb...let me read |
| 08:17.45 | linuxbangalore | because the phone is not yet registered. |
| 08:18.01 | linuxbangalore | what is the sip phone you are using? |
| 08:18.11 | muther | Ekiga, a sip phone for linux |
| 08:18.27 | muther | looks exactly like gnomemeeting |
| 08:18.45 | x86 | muther: that's because it IS gnomemeeting |
| 08:18.57 | x86 | muther: ekiga is the new name for gnomemeeting |
| 08:19.13 | muther | yeh I figured that |
| 08:19.58 | Libila | Anyone here use sellvoip as their provider? |
| 08:21.07 | muther | but I still can't get it to register with my asterisk server :) |
| 08:21.31 | SoftIce | Stephnie: how is your SIP.conf setup? |
| 08:21.41 | SoftIce | is it listening to your IP or is it set to use 0.0.0.0 |
| 08:21.48 | SoftIce | as 0.0.0.0 should bind to all IP's |
| 08:22.03 | Stephnie | yes it is 0.0.0.0 |
| 08:22.09 | Stephnie | ok I got your point... |
| 08:22.42 | SoftIce | Stephnie: i'm just giving you a way out |
| 08:22.47 | *** join/#asterisk bigjb (n=nbigjb@195.60.10.114) |
| 08:22.51 | Stephnie | but is there a way to do it through asterisk? other than using forwarding and proxy etc? |
| 08:22.54 | SoftIce | why not speak to your carrier and ask them how they advise you to set this up |
| 08:23.13 | SoftIce | Stephnie: I wouldn't believe your carrier would expect you to use such complex networking setup |
| 08:23.20 | Stephnie | thats the what I am gonna do if Its not possible with asterisk |
| 08:23.22 | SoftIce | and I do believe that 0.0.0.0 should take care of it all |
| 08:23.44 | x86 | SoftIce: Stephnie doesnt have both IPs locally |
| 08:23.49 | x86 | SoftIce: her carrier does |
| 08:24.00 | SoftIce | oohhhhhh |
| 08:24.03 | x86 | SoftIce: so Stephnie binding to local 0.0.0.0 will not fix the issue |
| 08:24.10 | SoftIce | yes, I see |
| 08:24.14 | SoftIce | sorry I thought she had 2 ip's |
| 08:24.24 | Stephnie | :) |
| 08:24.31 | x86 | her carrier does, 1 for SIP, 1 for RTP |
| 08:24.36 | SoftIce | Stephnie: yes, well then my theory with forwarding will work |
| 08:24.37 | Stephnie | yep... |
| 08:24.39 | x86 | which is totally retarded ;) |
| 08:24.47 | Stephnie | he he |
| 08:25.01 | x86 | Stephnie: easier to find another carrier most likely ;) |
| 08:25.11 | Stephnie | x86: retarded what?? SoftIce's theory or my carrier ? ;) |
| 08:25.25 | SoftIce | Stephnie your carrier! |
| 08:25.29 | x86 | carrier requiring SIP and RTP going to two different IPs |
| 08:25.58 | SoftIce | Stephnie: like x86 said, find another carrier, unless you comforatble with routing / packet filtering |
| 08:26.19 | x86 | especially layer7 filtering |
| 08:26.19 | SoftIce | if you using linux look at iptables if you using bsd look at pf |
| 08:26.19 | Stephnie | SoftIce: I believe your theory could work but I think I should talk to my carrier... |
| 08:29.00 | x86 | SoftIce: she'll need the layer7 filtering patches and experimental kernel support for l7 filtering to handle RTP on dynamic ports |
| 08:30.25 | *** join/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net) |
| 08:32.11 | SoftIce | x86: I don't see that getting pulled of that easy without extensive knowldge of firewalls |
| 08:37.25 | muther | could somebody look at the sip debug message and mabye give me a hint as to why i can't connect to my asterisk? http://pastebin.ca/191721 |
| 08:38.45 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:45.23 | LakeSolon | Trying to dig up some info and having a little more trouble than I thought (it's been a little while since I've played with Asterisk)... |
| 08:46.04 | LakeSolon | Is there a mechanism in IAX2 to receive a call, do a voice menu or whatever, and then tell the 'calling IAX device to forward to a different number? |
| 08:46.25 | LakeSolon | and if so, does any PSTN termination provider actually support something like that? |
| 08:46.38 | LakeSolon | As opposed to entering the number to forward to via their web interface. |
| 08:46.49 | LakeSolon | let's call it, 'client side conditional forwarding'. |
| 08:47.11 | LakeSolon | without having to send both the incoming and outgoing voice data over the same client internet connection. |
| 08:48.55 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:49.39 | hank | ARGHHH |
| 08:49.57 | hank | i just had to say that... |
| 08:50.14 | LakeSolon | It needed to be said. |
| 08:51.04 | hank | im glad im not the only one thinking so :) |
| 08:56.39 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 08:56.54 | flackes | Hello there |
| 08:57.18 | flackes | is there any one that can help me with Asterisk BLF and GXP - 2000 |
| 08:57.53 | *** join/#asterisk oQPa (i=name@237.Red-83-44-33.dynamicIP.rima-tde.net) |
| 08:59.09 | flackes | oO |
| 09:03.07 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:17.21 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:20.35 | edwar64896 | flackes: yeah - got that working here. |
| 09:36.52 | *** join/#asterisk kronic (n=blowfish@static-203-87-64-28.vic.chariot.net.au) |
| 09:48.57 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 09:50.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:56.39 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 09:56.42 | flackes | Hello there |
| 09:56.57 | flackes | im looking for some help with a GXP - 2000 and busy line filter |
| 09:57.07 | flackes | i have everything working except pickup of exturnal calls |
| 09:57.15 | flackes | i seem to get a 603 error and cant work out why |
| 09:57.32 | flackes | any one that can help |
| 09:59.41 | flackes | nice and quite :P |
| 10:01.50 | *** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-10.perspektivbredband.net) |
| 10:02.13 | Curus | Is it possible to issue multiple asterisk commands at once with asterisk -rx ? |
| 10:02.49 | flackes | not something i have done |
| 10:02.59 | flackes | but i belive you can only send one command at a time |
| 10:03.08 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:03.08 | Curus | Oh well |
| 10:04.43 | flackes | Any one got BLF working? |
| 10:04.50 | Curus | Yes |
| 10:05.00 | flackes | i keep getting a 603 error |
| 10:05.07 | flackes | for picking up exturnal lines |
| 10:05.15 | flackes | the rest works fine |
| 10:05.19 | Curus | Hmm, asterisk -r doesn't like having stdin redirected |
| 10:05.24 | Curus | Which phone? |
| 10:05.29 | flackes | grandstream |
| 10:05.43 | Curus | Grandstream has its very own way to do call pickup |
| 10:05.44 | flackes | gxp - 2000 |
| 10:05.52 | flackes | tell me about it |
| 10:05.59 | Curus | You need to put in something that reacts to **<number> |
| 10:06.07 | flackes | i have done that |
| 10:06.27 | flackes | and it works for picking up sip calls ect |
| 10:07.14 | flackes | [BLF_group_pickup] |
| 10:07.14 | flackes | include =>inbound-from-stem |
| 10:07.14 | flackes | exten => _**.,1,NoOp(${EXTEN}) |
| 10:07.14 | flackes | exten => _**.,2,Pickup(${EXTEN:2}) |
| 10:07.14 | flackes | exten => _**.,3,Hangup |
| 10:07.26 | Curus | Does asterisk react at all when you press the pickup button? |
| 10:07.43 | Curus | As in, something gets written when you're looking at asterisk -rvvvvvv |
| 10:07.51 | flackes | https://support.voiptalk.org/bugtracker/view.php?id=6955 |
| 10:08.00 | flackes | that is my bug :P |
| 10:08.10 | flackes | been working on this for ages now :( |
| 10:08.17 | flackes | and i have the verbose set to 10 |
| 10:08.39 | Curus | I get a login prompt |
| 10:08.59 | flackes | all sterisk reports is |
| 10:08.59 | flackes | <PROTECTED> |
| 10:08.59 | flackes | <PROTECTED> |
| 10:09.27 | flackes | [default] |
| 10:09.27 | flackes | include => stem |
| 10:09.27 | flackes | include => to-siemens |
| 10:09.27 | flackes | include => BLF_Group_1 |
| 10:09.27 | flackes | include => BLF_group_pickup |
| 10:09.28 | flackes | [inbound-from-stem] |
| 10:09.30 | flackes | include => internal |
| 10:09.32 | flackes | include => DefExt |
| 10:09.34 | flackes | include => voicemail |
| 10:09.36 | flackes | include => outbound |
| 10:09.38 | flackes | include => BLF_group_pickup |
| 10:09.40 | flackes | include => BLF_Group_1 |
| 10:09.42 | flackes | include => clfwd |
| 10:09.44 | flackes | ;Test section for BLF on Grandstreams for Stem |
| 10:09.46 | flackes | [BLF_group_pickup] |
| 10:09.48 | flackes | exten => _**XXXX,1,Pickup(${EXTEN:2}) |
| 10:09.50 | flackes | exten =>_**XXXX,2,Hangup |
| 10:09.52 | flackes | [BLF_Group_1] |
| 10:09.54 | flackes | exten =>7000,hint,SIP/7000 |
| 10:09.56 | flackes | exten =>7001,hint,SIP/7001 |
| 10:09.58 | flackes | exten =>7002,hint,SIP/7002 |
| 10:10.00 | flackes | exten =>7003,hint,SIP/7003 |
| 10:10.02 | flackes | exten =>7004,hint,SIP/7004 |
| 10:10.04 | flackes | that is the kinda thing i have gone for |
| 10:10.06 | flackes | and this is what grandstream said |
| 10:10.08 | flackes | Dear Andrew Shelton, |
| 10:10.10 | flackes | "603" means "Decline", which means Asterisk does NOT accept "**extention" to pick up the call in a pickup group. |
| 10:10.13 | flackes | Our reference is just for your reference ONLY. It will NOT work if you copy that configuration. We are not supporting Asterisk programing. |
| 10:10.16 | flackes | Your Asterisk programing is the source of the problem. Please google wiki or user forum to get some help or hints. |
| 10:10.19 | flackes | Thanks. |
| 10:10.27 | flackes | the least they could do was point me in the right direction |
| 10:11.17 | Mw3 | ~pb |
| 10:11.21 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 10:11.40 | flackes | oO |
| 10:11.42 | flackes | sorry |
| 10:12.04 | flackes | any ideas? |
| 10:12.27 | Mw3 | no, i do not use grandstream hardware |
| 10:13.52 | flackes | not even like there is anywhere to give you an idea on how to get this to work |
| 10:16.42 | flackes | any one got an idea of somewher i can get some help? |
| 10:17.05 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 10:17.37 | X-Rob_ | flackes, you want 'exten => _**.,1,Pickup(${EXTEN:2}) |
| 10:17.43 | X-Rob_ | in whatever context the phones are in |
| 10:17.49 | Curus | Well, why would it work to pickup 7000? Is there a call at 7000? |
| 10:17.57 | *** part/#asterisk SoftIce (n=awk@vc-196-207-45-253.3g.vodacom.co.za) |
| 10:18.00 | tzafrir | hmmm, this pb factoid in itself is rather long. Is it used in any channels other than #asterisk ? |
| 10:18.15 | Curus | Err 7001 that is |
| 10:19.38 | flackes | 7001 is a sip phone |
| 10:20.57 | tzafrir | jbot, no pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:21.02 | jbot | tzafrir: okay |
| 10:21.06 | tzafrir | ~pb |
| 10:21.07 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:21.17 | tzafrir | ~pastebin |
| 10:21.18 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 10:21.32 | tzafrir | jbot, no pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:21.33 | jbot | okay, tzafrir |
| 10:21.33 | hank | paste.debian.net is imho the newer version of channel.debian.net/paste/ |
| 10:21.40 | tzafrir | ~pastebin |
| 10:21.41 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:21.53 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:24.53 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 10:27.20 | *** join/#asterisk type0h (i=type0@209-193-49-206-cdsl-rb1.anc.acsalaska.net) |
| 10:33.06 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 10:33.13 | flackes | nerf keep getting disconnected |
| 10:35.15 | flackes | http://channels.debian.net/paste/3944 |
| 10:35.17 | flackes | that is what i get |
| 10:38.53 | X-Rob_ | flackes, looks good to me. |
| 10:39.02 | X-Rob_ | if it's still not working, check /var/log/asterisk/full |
| 10:40.10 | flackes | http://channels.debian.net/paste/3945 |
| 10:40.24 | flackes | that is all the config i have done to get BLF to work |
| 10:40.28 | flackes | ok checking the log now |
| 10:43.04 | flackes | hmm dont seem to have that log file oO |
| 10:47.47 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 10:47.54 | flackes | how come i keep getting disconnected? |
| 10:48.45 | flackes | X-Rob_do i have to restart asterisk to get it to work or will a reload do? |
| 10:48.53 | flackes | i restart the phones to get them to re register |
| 10:49.20 | flackes | omg this suxs so bad... everything else works apart from the pickup |
| 10:49.47 | flackes | but dont understand why the grandstream would report a 603 error |
| 10:50.02 | flackes | surly if it cant find the code it would display a 408 error or something |
| 10:50.15 | flackes | but asterisk does not say anything |
| 10:51.07 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:51.09 | flackes | <PROTECTED> |
| 10:51.29 | flackes | that measn is spawning **7002 in inbound-from-stem correct? |
| 10:52.00 | flackes | but if im using exten => _**.,2,Pickup(${EXTEN:2}) |
| 10:52.14 | flackes | surly it should spawn 7002 in inbound-from-stem? |
| 10:52.23 | X-Rob_ | flackes, enable debug in /etc/asterisk/logger.conf |
| 10:52.30 | X-Rob_ | then do it |
| 10:52.34 | X-Rob_ | then look at the error log |
| 10:52.49 | flackes | ok... thanks so much for the help btw |
| 10:52.55 | flackes | been working on this for months |
| 11:01.09 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 11:01.18 | flackes | any way to stop me getting DC |
| 11:06.00 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:06.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:09.21 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 11:09.52 | MRH2 | anyone taken a polycom phone apart? |
| 11:16.43 | MRH2 | (trying to find out if the digit keys are easy to replace) |
| 11:16.56 | *** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com) |
| 11:17.03 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 11:17.14 | flackes | X-Rob_ would you mind having a quick look through the log file? |
| 11:17.32 | X-Rob_ | bit busy now |
| 11:17.39 | X-Rob_ | search for 'Pickup' and read why it's not working |
| 11:17.52 | jgoo | flackes: actually, good point, on trixbox, which is the best way to view the log file? via ssh I guess right? |
| 11:17.59 | flackes | X-Rob_ok..well i dont mind waiting for a time that is best for you |
| 11:18.25 | flackes | im using SSH and FTP |
| 11:18.40 | flackes | but im no way the best |
| 11:18.49 | flackes | only been doing this 3 months |
| 11:19.26 | jgoo | is there somewhere that lists the locations of all the relevant logs... |
| 11:19.41 | flackes | Oct 5 12:12:51 DEBUG[7723] chan_sip.c: build_route: Contact hop: <sip:7003@192.168.1.94:5060> |
| 11:19.41 | flackes | Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing NoOp("SIP/7003-b721be28", "**7002") in new stack |
| 11:19.41 | flackes | Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing Pickup("SIP/7003-b721be28", "7002") in new stack |
| 11:19.41 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found. |
| 11:19.42 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible... |
| 11:19.54 | flackes | no originating channel oO |
| 11:20.25 | jgoo | flackes: where is the location of that log file? |
| 11:20.41 | flackes | all the logs should be in /var/log/asterisk |
| 11:20.47 | jgoo | cheers |
| 11:20.54 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-55-82.phlapa.east.verizon.net) |
| 11:23.00 | flackes | ok so call puckup is not possible because it cant find its originating channel... now im just really confused |
| 11:23.10 | flackes | puckup = pickup |
| 11:24.10 | flackes | any one else have problems with grandstream GXP -2000 and Asterisk Pickup? |
| 11:26.01 | jgoo | erp. I am in asterisk CLI, verbosity 9 |
| 11:26.11 | flackes | yes |
| 11:26.18 | jgoo | I try and register with xlite, but NOTHING comes out on the asterisk side |
| 11:26.20 | flackes | X-Rob_what time you think you would be free? |
| 11:26.33 | flackes | using IAX? |
| 11:26.34 | jgoo | yesterday I had no problems with xlite, now it gives me that 408 timeout |
| 11:26.36 | *** join/#asterisk ernie_ (i=jgeraert@193.202.9.42) |
| 11:26.46 | jgoo | flackes: no, SIP (I think, heh, no I am sure) |
| 11:26.55 | flackes | what file u editing? |
| 11:27.34 | jgoo | I kinda added a new inbound, 7777, that just pointed to an IVR, then is stopped working... I also renamed from-zaptel to from-pstn a few places, but then |
| 11:27.43 | jgoo | I think I renamed them back :s (didn't backup :( was hacking) |
| 11:28.09 | jgoo | does trixbox install customize the out-of-the-box config files? or can I redownload them? |
| 11:28.33 | flackes | i dont use a trixbox |
| 11:28.36 | flackes | so would not know |
| 11:28.49 | flackes | but the chances are you have just put something in the wrong place |
| 11:29.46 | jgoo | yeah... one of the from-pstn must be still changed... |
| 11:29.59 | *** join/#asterisk liran_ (n=liran@212.199.177.208.static.012.net.il) |
| 11:30.19 | jgoo | does help this office is so damn noisy... I got more done hacking at home... >.< grrr. |
| 11:30.26 | flackes | lol |
| 11:30.35 | flackes | its all quite here |
| 11:30.37 | flackes | just stuck |
| 11:30.53 | flackes | and i dont see why pickup cant find its originating channel |
| 11:32.14 | *** part/#asterisk liran_ (n=liran@212.199.177.208.static.012.net.il) |
| 11:33.16 | flackes | X-Rob_dont mind waiting all day :P |
| 11:33.20 | jgoo | damn, yesterday I had no problem with this 408, then it just happened... restart... no avail... |
| 11:33.50 | flackes | its a extensions problem, i guess |
| 11:33.53 | flackes | what have you added |
| 11:34.24 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:38.40 | *** join/#asterisk Zouzou (n=test@mail.splendor.net) |
| 11:39.15 | Zouzou | What does Asterisk only supports g723.1 pass-thru means? |
| 11:39.29 | ernie_ | it cant recode to another codec |
| 11:40.23 | ernie_ | but it can pass it on to another device that supports g723 |
| 11:40.37 | Zouzou | what do u mean by recode to another codec |
| 11:41.12 | ernie_ | it cant translate from g723 to g711 |
| 11:41.15 | ernie_ | or to gsm |
| 11:41.21 | Druken | do you understand what a codec is? |
| 11:41.23 | ernie_ | or to whatever other codec |
| 11:41.29 | Zouzou | yes |
| 11:42.02 | ernie_ | show translation |
| 11:42.18 | ernie_ | then you see a matrix of translation times between the codecs |
| 11:42.48 | Zouzou | so if i have to use i need that all mt phones are g723 capables |
| 11:43.56 | ernie_ | or use another codec |
| 11:44.15 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 11:44.29 | flackes | any one that can help with a Grandstream GXP-2000 and BLF problem? |
| 11:44.31 | Zouzou | if i mentioned many codecs in allow=codecname in which sequence does asterisk us them? |
| 11:44.55 | flackes | the order you put them oO |
| 11:44.56 | ernie_ | in the order you mention them |
| 11:45.10 | Zouzou | ok , thanks a lot guys |
| 11:45.21 | flackes | some one has to be a guru :P |
| 11:45.31 | flackes | help the man about to have no hair |
| 11:48.20 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 11:48.24 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 11:51.10 | *** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 11:51.48 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 11:51.52 | brif8 | Hi all, can one simply delete /var/lib/asterisk/astdb or is there a correct way to clean and refresh the asterisk database ? |
| 11:55.35 | RoyK | erm.. http://www.asterisk.org/node/99 describes 'maxauthreq' option to combat DoSing IAX2, and asks admins to set this to a 'reasonable value'. wtf is a reasonable value? 10? 10000? |
| 11:57.14 | *** join/#asterisk vetaly (n=vetaly@tmp.megalink.com.ru) |
| 11:57.26 | *** part/#asterisk vetaly (n=vetaly@tmp.megalink.com.ru) |
| 11:58.42 | kronic | has anyone used asterisk in a call centre environment, and can recommend a method for obtaining reports etc...? |
| 11:58.52 | kronic | is using the CDR log sufficient? |
| 11:59.33 | brif8 | kronic: in most cases, what are you seeking ? |
| 12:00.18 | kronic | well, more or less a wealth of data, so that it can be manipulated to produce details about agents, queues and groups |
| 12:00.32 | RoyK | perhaps queue_log |
| 12:00.59 | RoyK | there reallly should be a good combined queue_log and cdr log |
| 12:01.33 | kronic | cheers |
| 12:02.07 | brif8 | if using queues and agents you may just want the queue_log if you're seeking call data all is logged into CDR. or you could parse through messages even asterisk stat CDR is a nice tool if you store your CDR in a database |
| 12:02.26 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 12:03.11 | brif8 | http://areski.net/asterisk-stat-v2/about.php check it out |
| 12:03.14 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 12:03.39 | flackes | any reason i keep getting DisconnecteD? |
| 12:03.53 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 12:04.30 | kronic | thanks mate |
| 12:05.00 | flackes | any time |
| 12:05.10 | flackes | Any one with some BLF exsperiance |
| 12:05.14 | flackes | with grandstream |
| 12:05.31 | flackes | think i just dont have my code in the correct place.. |
| 12:06.06 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 12:06.37 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:06.51 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:06.52 | kronic | our provider basically provides that information to us |
| 12:06.59 | kronic | it looks interesting though |
| 12:07.05 | flackes | ? |
| 12:07.20 | kronic | reply to brif8 though |
| 12:07.32 | flackes | sorry im confused :P |
| 12:07.40 | flackes | got DC |
| 12:10.02 | flackes | so any one :P |
| 12:11.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:13.07 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:14.44 | flackes | Grandstream :P |
| 12:15.31 | Rhizome | eww! |
| 12:15.34 | Rhizome | snom ;) |
| 12:15.47 | flackes | i dont get a choice |
| 12:15.52 | flackes | need to get BLF to work |
| 12:23.33 | brif8 | flackes: I think that BLF only needs the subscribecontext = in Sip.conf as far I remember (don't use grandstreams) |
| 12:25.53 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 12:26.07 | brif8 | www.grandstream.com.cn/download/other/FAQ_and_Example_for_Asterisk_Configuration_for_GXP-2000.pdf |
| 12:26.13 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 12:27.25 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.42) |
| 12:29.21 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:33.00 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 12:33.19 | flackes | omg at the [13:34] * [10053] Software caused connection abort |
| 12:39.01 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 12:40.10 | sudhir492 | Hi All |
| 12:42.48 | brif8 | Anyone using a FXO gateway (Clipcomm CG 410 would be nice) I can't even get registration to work SIP SHOW PEERS has "3000/3000 192.168.0.39 D 5060 UNREACHABLE" UNREACHABLE ?? Yet it is on the same subnet |
| 12:43.38 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:44.31 | pablus | morning |
| 12:49.10 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 12:49.56 | sudhir492 | brif8: I use FXO cards with Asterisk |
| 12:50.21 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:51.15 | brif8 | sudhir492: I have the TDM400p with 1 FXS and 2 FXO but I get bad quality, so I'm trying an external gateway to isolate the problem. I have tried everything for quality even down to fxotune and many others :( |
| 12:52.04 | devel | so.... anybody here with an FXO audiocodes (analog) and working inbound callerid to it? |
| 12:53.17 | *** join/#asterisk oriso (n=oriso@67.71.244.174) |
| 12:53.57 | sudhir492 | brif8: Where are you using. I am in Virginia, USA and just installed an Asterisk with 2 TDM400, all 8 ports FXO and have had no problem at all |
| 12:55.23 | sudhir492 | I used a a Compaq PC (Sempron 3400+), 512 MB of Memory that was on sale for $327. |
| 12:55.45 | sudhir492 | What kind of problem are you having? |
| 12:56.03 | *** join/#asterisk mtaht4 (n=m@h-72-244-145-197.phlapafg.covad.net) |
| 12:56.51 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:57.56 | sudhir492 | brif8: The problem could be in FXS. Also depending on the phone you are using, the problem could get worse. If you can, do not use FXS, instead try a VoIP phone |
| 13:00.23 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:01.44 | *** join/#asterisk sting3r (n=sting3r@63.99.54.130) |
| 13:02.27 | brif8 | sudhir492: I'm in Florida, USA, I have an Intel 2 GHz 2 GB Ram PC. I have a mix between VoIP IP Phones (snom 200) and std. analog. My boss just loves a cordless phone. I have the wire running from the PSTN box to the FXO port on the TDM and then from the FXS port to the old jacks around the office. There is a major static HISS and crackle that gets worse during lat afternoon and into the evening. Sometimes you can't hear the other party at all. |
| 13:02.58 | brif8 | It will come and go at will / random the call might start great have a small hiss go great again the get so bad you can't hear and it will cut you off, click |
| 13:03.18 | RoyK | the land of the free that bans children's books like harry potter? :D |
| 13:03.45 | cpm | brif8, what do you get when jack the basestation directly into the fxs port? |
| 13:03.52 | cpm | <PROTECTED> |
| 13:04.23 | brif8 | cpm same problem |
| 13:05.36 | cpm | and if you don't use the wireless? |
| 13:06.45 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 13:06.46 | brif8 | That was Sprint's trick use a wired phone. It improves somewhat but it is still there |
| 13:08.08 | brif8 | and like I said it is random some calls are fine other just give up, which for business if very bad and my neck on the boss's block |
| 13:08.21 | [TK]D-Fender | brif8: If you reboot the box, does the static go away? |
| 13:09.00 | brif8 | no |
| 13:09.03 | *** join/#asterisk rados_ (n=rados10@c-68-62-71-76.hsd1.mi.comcast.net) |
| 13:09.03 | [TK]D-Fender | brif8: Also, if you don't get static on the SIP hardphones, then simply get an ATA and ditch Zaptel FXS. Its ass anyways... |
| 13:09.38 | brif8 | Well I was trying with an FXO first, because IP Phone to FXS works ok |
| 13:10.21 | brif8 | thus I bought a Clipcomm CG 410 (4 x FXO ports) but I can't get the dumb thing to work with Asterisk. I have just now in the last 20 seconds found qualify = no helps the registration side |
| 13:10.26 | [TK]D-Fender | brif8: Ok, so if SIP-FXO is good, SIP-FXS is good, and FXS-FXO is bad well.... guess you'll want to ditch the FXS |
| 13:10.56 | cpm | indeed, get something that works |
| 13:11.16 | brif8 | :) |
| 13:11.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.09 | rados_ | Hello everyone; I'm trying to connect Asterisk to Avaya Definity. I'm able to do that but I have trouble hearing audio from Asterisk on the Avaya IP Phone. Can anyone suggest how I can fix that? |
| 13:14.06 | *** join/#asterisk oriso (n=oriso@67.71.244.174) |
| 13:15.30 | brif8 | [TK]D-Fender: Ok why then if SIP -FXS is good I don't use much SIP-FXO and FXS-FXO is not good. How does it become FXS to be ditched ? |
| 13:16.11 | anonymouz666 | it's been long time lonely lonely lonely time |
| 13:16.11 | oriso | Hello all! I have a small question I hope one of you can help answer. We have 8 analog lines for incomng and outgoing external calls. Right now, our telco is sending calls from line 1 to 8, if they're busy. Asterisk is also using line 1 to 8, when connecting for outgoing calls, resulting in not so funny cross-lines mish mashing when say, a call comes in just as a user attempts an outgoing call. Is there a way to tell asterisk to start using lines 8 to 1 inst |
| 13:17.21 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-118215.otenet.gr) |
| 13:17.33 | [TK]D-Fender | brif8: Because you have fewer FXS ports than FXO. Also Zaptel FXS is more expensive and less functional than ATA FXS. Add to that the complexity of configuring that gateway that has more ports than you needed and a higher cost. You are working on "the hard way" now with little benifit. You'd have been better off spending $70 and getting a nice litle ATA and making your boss happy DAYS ago. |
| 13:17.41 | [TK]D-Fender | brif8: ... in short :) |
| 13:18.49 | brif8 | [TK]D-Fender: ok. but if I get the FXO working then we can expand and receive multple incoming calls (provided I get the thing to work) |
| 13:19.05 | brif8 | I'll go get an ATA to swap out the FXS np. |
| 13:19.19 | Mw3 | oriso: asterisk does not use busy channels for dialing out |
| 13:19.21 | [TK]D-Fender | brif8: Which FXO? |
| 13:20.57 | jtexter3 | oriso: I believe in your dial statement you use a captial g, so change Dial(Zap/g1/${EXTEN}) to Dial(Zap/G1/${EXTEN}) |
| 13:21.07 | [TK]D-Fender | oriso: Strike Mw3's comment for a sec. You set your lines into the same group in zapata.conf, and which order they pull in depend on how you Dial them. G1 goes in one direction, g1 in the other (capitalizateion) |
| 13:21.13 | brif8 | [TK]D-Fender: the FXO gateway (clipcomm CG 410) plus the two FXO on the TDM400p would give us a total of 6 possible concurrent calls |
| 13:22.33 | [TK]D-Fender | brif8: Personal suggestion : Ditch them both and get a Sangoma A200. Scale up cheaper than those 2 being put into a mish-mash setup. Because they are different techs you can't pick a resource with a simple Dial staament (can't pool your channels). Don't do it. |
| 13:23.35 | [TK]D-Fender | brif8: Never make normal lines span tech or devices (multiple gatways) |
| 13:23.53 | brif8 | ok |
| 13:23.57 | brif8 | thanks |
| 13:25.18 | pif | hi, where can I find detailed info on zaptel.conf's "span=" declarations? |
| 13:25.31 | pif | I have a TE410P |
| 13:25.43 | brif8 | Is there anyway to refresh the astdb (clear it out and start again ? ) |
| 13:25.45 | *** join/#asterisk wahjava (n=admin@unaffiliated/wahjava) |
| 13:25.58 | wahjava | hi channel |
| 13:26.21 | wahjava | is it possible to use external modem with asterisk |
| 13:26.23 | wahjava | ?? |
| 13:26.28 | [TK]D-Fender | brif8: Its just a file. Go delete it and it will get rebuilt |
| 13:26.39 | [TK]D-Fender | wahjava: Not as a normal line. |
| 13:26.42 | *** join/#asterisk StiXantti (n=Antti@81.29.128.60) |
| 13:26.49 | [TK]D-Fender | pif : Read the book, or the WIKI |
| 13:26.51 | [TK]D-Fender | ~book |
| 13:26.52 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:26.53 | [TK]D-Fender | ~wikis |
| 13:26.55 | jbot | methinks wikis is http://www.voip-info.org |
| 13:27.07 | StiXantti | Just bought the book =) |
| 13:27.22 | StiXantti | But I have a BIIIG problem... |
| 13:27.39 | [TK]D-Fender | StiXantti: www.drphil.com |
| 13:27.41 | StiXantti | A part of the extensions are unreachable |
| 13:27.43 | wahjava | [TK]D-Fender: okay |
| 13:27.52 | pablus | morning |
| 13:27.53 | [TK]D-Fender | StiXantti: Ok, some details please.... |
| 13:28.07 | StiXantti | ...as some work just fine... |
| 13:28.14 | [TK]D-Fender | sti, I'd suggest putting everything up in www.pastebin.ca for us to see |
| 13:29.05 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:29.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:29.15 | StiXantti | Looking at the logs, it seems that Asterisk is not receivin the answers from the phone in time |
| 13:29.23 | *** part/#asterisk wahjava (n=admin@unaffiliated/wahjava) |
| 13:29.47 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:29.48 | [TK]D-Fender | StiXantti: please paste the output and config files in question if you wish us to try and help you. |
| 13:29.55 | StiXantti | D-Fender: I'll check that bin now |
| 13:29.59 | muppetmaster | Hello all. |
| 13:30.46 | oriso | thanks jtexter3 and [TK]D-Fender, I'll try that as soon as I can. |
| 13:30.47 | muppetmaster | I am having a problem with app_queue locking after about 15 agents, in the same way as in 6626 which says the patch to fix it was applied to v1.2.12.1 (which we are using, and I verified in looking at chan_agent.c) that it was there. |
| 13:30.48 | jgoo | guy I am confused with from-pstn and from-zaptel , when should you use which and where? :s |
| 13:31.51 | muppetmaster | Problem is, the issue persists. And looking at the notes, it appears another patch might be needed, but I have no idea where it may have been 'uploaded' to. The patch is manager_eventq_backport-1.2.10.patch, any ideas where it would be uploaded to if the guy in the notes of the bug on Mantis just said I 'uploaded'? |
| 13:32.16 | muppetmaster | And could that patch be applied to v1.2.12.1 anyway? |
| 13:32.27 | [TK]D-Fender | jgoo: Depends where you chose them and why. |
| 13:32.29 | lilalinux | If I have 2 extensions of which the first is a prefix of the second, will the second be used after the end of the first? |
| 13:32.35 | muppetmaster | More details here: http://bugs.digium.com/view.php?id=6626 |
| 13:32.55 | rados_ | can anyone point me to any resource on connecting Asterisk to Avaya Definity as an endpoint? |
| 13:33.02 | rados_ | I would really appreciate it. |
| 13:33.05 | jgoo | [TK]D-Fender I was reading some docs that says you may need to change them... erm, I have a pretty new install, and I am trying to get incoming and outgoing, on my last trixbox, it seemed to just work (but I was developing at home, much more relaxed) |
| 13:33.25 | [TK]D-Fender | jgoo: So you're using Trixbox? |
| 13:33.46 | brif8 | [TK]D-Fender, can you simply delete /var/lib/asterisk/astdb and have asterisk rebuild it again ? |
| 13:33.54 | jgoo | I have a TDM04B, 4 fxs ports, one, left most, has a phone line in it... (yes, but this isn't a trixbox question, I am editing the conf file...) |
| 13:33.54 | [TK]D-Fender | brif8: Yup. |
| 13:34.36 | StiXantti | [TK]D-Fender: can't really paste since it's in production and the boss... well You propably get the idea |
| 13:34.38 | *** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 13:34.38 | jgoo | but if you want I could go into #freepbx, but this is really an asterisk Q, the last time i set this up it wasn't using trixbox, just compile from SVN, using suse 10.1 |
| 13:34.52 | [TK]D-Fender | jgoo: You're not supposed to be manually editing that file. its generated by FreePBX and will get blown away and since you are asking which contexts to use, that is very exclusively a FreePBX question. |
| 13:35.02 | jgoo | ah ok |
| 13:35.22 | jgoo | now I get it... ut I am editing it through the phpconfig app... anyway |
| 13:35.36 | [TK]D-Fender | jgoo: Contexts do what you tell them to, and since those ones are built by FreePBX and not YOU, I guess you see where this is going... and hopefully WHY. |
| 13:36.08 | jgoo | ok, I have to read up more on contexts and the structure of the confs... I find them very hard to follow right now as I don't know the logci |
| 13:38.09 | [TK]D-Fender | jgoo: Well this is FreePBX logic, and the prime reason nobody here wants to touch it. Why are you modifying contexts directly in the first place? |
| 13:38.51 | mosty | is the bristuff patch only needed for BRI hardware? |
| 13:39.11 | StiXantti | [TK]D-Fender: the main problem (I think) is that Asterisk leaves pipes open or something. |
| 13:39.26 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 13:39.36 | [TK]D-Fender | mosty: There is a DEVSTATE patch that is part of BRISTUFF that can be useful for Presence manipulation, but no its not at all "necessary". |
| 13:40.04 | [TK]D-Fender | StiXantti: Could you give a more specific example and show some backup for it? |
| 13:40.27 | StiXantti | jgoo: recently installed trixbox and did the same thing. Quite hard to follow all the "xxx_additional" files and extensions =( |
| 13:40.50 | StiXantti | [TK]D-Fender: I'll try to paste something... |
| 13:42.01 | [TK]D-Fender | FreePBX and all other GUI's are a dead-end upon which you should give up hopes of "modifying" for the most-part. Its built to turn * into a cookie cutter system, and if you don't like the shape of your cookie, TFB. |
| 13:43.03 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 13:44.31 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 13:44.56 | kink0 | hello |
| 13:44.58 | [TK]D-Fender | And the worst part is when people keep thinking their problem is * and not the GUI...... LOL! Its the user's fault for lack of understanding the GUI or a bug in the way the GUI does its job. Either way, nothing we want to hear about here. |
| 13:45.16 | StiXantti | [TK]D-Fender : http://pastebin.ca/191895 --- Asterisk log # 1 |
| 13:45.23 | devel | yes, i'd like to publicly decry GUI |
| 13:45.46 | sudhir492 | Anyone using Polycom phones here? |
| 13:46.32 | devel | now, [TK]D-Fender, if we could only explain that to the bosses in words they understand |
| 13:47.01 | devel | i'll start by quoting you :) |
| 13:47.10 | StiXantti | [TK]D-Fender : http://pastebin.ca/191899 --- X-lite log # 1 |
| 13:49.04 | [TK]D-Fender | devel: I'm imminently quotable, and my works are largely public domain. Have fun :) |
| 13:49.35 | [TK]D-Fender | sudhir492: Plenty of us, just ask your question. |
| 13:49.50 | StiXantti | And like I said in the first place - not all phones (extensions) are misbehaving |
| 13:50.35 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:52.32 | *** join/#asterisk Holos (n=asdf@204.101.26.106) |
| 13:52.37 | [TK]D-Fender | StiXantti: HRm... a lot of "stuff" in there.... |
| 13:52.59 | StiXantti | And furthermore, "sip show peer xxxxxxx" seems ok, but when calling to this extension it goes to announcement in about ten secs "Out of service - check the nr" |
| 13:53.10 | Holos | What command do I use in the dial plan to wait for the user to enter digits, and proceed once they enter the pound key? |
| 13:53.13 | StiXantti | [TK]D-Fender: Yes - I know =( |
| 13:53.46 | [TK]D-Fender | Holos: "show application read" |
| 13:54.06 | StiXantti | [TK]D-Fender: Just so happens - I just got through! Maybe Digium informed my Asterisk that I bought the book.... ....scary =) |
| 13:54.33 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:54.48 | sudhir492 | How do you get Paging to work with Polycom 501 phones |
| 13:54.51 | [TK]D-Fender | StiXantti: Wait a sec... you are talking to a NAT'd client? |
| 13:55.11 | [TK]D-Fender | sudhir492: Look at "polycom paging" on the WIKI. its tells you what you need to do. |
| 13:55.23 | Holos | [TK]D-Fender - Thanks.. I knew it was there, but couldn't remember what it was. I'm in the process of re-writing the "Follow Me" script for portable extensions and it has a few commands that have changed. |
| 13:55.33 | pif | [TK]D-Fender : thanks for the pointers, but there is nothing in the book or wiki about span= declarations |
| 13:56.02 | sudhir492 | I did look at the wiki, but my polycom phone is not answering for some reason. It keeps on ringing and ringing |
| 13:56.07 | pif | span=1,0,0,ccs,hdb3 |
| 13:56.18 | pif | for instance is explained nowhere |
| 13:56.20 | [TK]D-Fender | pif : sure there is, and if you make me go look and provide the link (which I'm likely to find in < 1 minute) you know I'm going to have to smack you upside the head right? (no I don't jsut have a link sitting around) |
| 13:56.30 | Holos | sudhir492: Did you set the Alert Info? and set the ring type to be Ring_Ans? |
| 13:56.32 | [TK]D-Fender | :) |
| 13:56.59 | sudhir492 | Yes, Idid |
| 13:57.18 | Holos | sudhir492: In your Sip.cfg set: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans" voIpProt.SIP.alertInfo.2.class="4"/> |
| 13:57.53 | [TK]D-Fender | pif : http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax |
| 13:58.09 | pif | ahh, nice :) |
| 13:58.09 | Holos | sudhir492: Then call it with a exten => _2XX,1,Set(_ALERT_INFO="Ring_Ans") |
| 13:58.09 | [TK]D-Fender | pif : See? Sub 1 minute results. You aren't trying very hard.... |
| 13:58.11 | lilalinux | how do I configure the outgoing msn when dialing out with CAPI/ISDN1/... ? |
| 13:58.23 | pif | I was looking at the zaptel.conf page |
| 13:58.47 | [TK]D-Fender | pif : open your eyes an JFGI |
| 14:00.21 | pif | dict, No definitions found for "JFGI" |
| 14:03.18 | *** join/#asterisk pjv (n=pjv@cor8-ppp3839.bur.dialup.connect.net.au) |
| 14:04.06 | [TK]D-Fender | ~jfgi |
| 14:04.08 | jbot | rumour has it, jfgi is http://www.justf*ckinggoogleit.com/ |
| 14:04.38 | devel | come on, jbot, the "*" isn't valid in URLs! |
| 14:05.08 | Nugget | google blocked that site anyway. it was a sad day. |
| 14:05.13 | Nugget | that site ruled. |
| 14:05.14 | [TK]D-Fender | You'd be amazed and how many answers I give out not having known the topic matter previously. I'm just capable of finding what anyone with an IQ higher than Lassie should be able to and spit it back for you. |
| 14:05.30 | Nugget | [TK]D-Fender is our resident google proxy. |
| 14:05.45 | [TK]D-Fender | Nugget: That, and Polycom God ;) |
| 14:05.50 | Nugget | heh |
| 14:06.00 | *** join/#asterisk knhor (n=knhor@cpe-70-125-158-177.satx.res.rr.com) |
| 14:06.51 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-192.dhcp.southerncoastalcable.net) |
| 14:06.59 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 14:07.26 | QbY | any ideas why # transfers won't work. the dial command has a 't' in it, and in features.conf blindxfer => # |
| 14:07.34 | Holos | [TK]D-Fender: Can you tell me how to jump priorites when a DB read fails to find a key? |
| 14:08.37 | [TK]D-Fender | Holos: It doesn't, and you shouln't try to. Priority jumping is DEAD. Use the new functions. |
| 14:08.47 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:09.14 | [TK]D-Fender | Holos: And there is a specific one to check the existance of a keyt (not just test if the attempted read comes back "blank") |
| 14:12.41 | Holos | [TK]D-Fender: So I guess I should use GotoIF(${$DB(portable/${targetpn}) and jump to my two places depending on if it exists or not... |
| 14:13.50 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 14:14.00 | [TK]D-Fender | Holos: Pretty much... |
| 14:14.00 | flackes | razu_ |
| 14:14.07 | flackes | mt |
| 14:14.21 | flackes | any one that has used GXP-2000 and asterisk and got BLF to work? |
| 14:14.32 | [TK]D-Fender | Holos: Actually you should only jump to 1 place, and simply continue on for the other. |
| 14:14.41 | razu | flackes : ? |
| 14:14.46 | [TK]D-Fender | flackes: Plenty of people, and its well documented on the WIKI |
| 14:14.47 | flackes | miss tell sorry |
| 14:15.03 | flackes | wiki.....dont seem to be able to find it |
| 14:15.14 | flackes | must be blind |
| 14:15.28 | flackes | just cant get the **.Pickup to work |
| 14:16.14 | flackes | [TK]D-Fender think you could point me in the right direction |
| 14:17.41 | knhor | flackes: you need to use "hint"s |
| 14:17.58 | flackes | yea i have all that working |
| 14:18.04 | flackes | all the lights work ect |
| 14:18.24 | flackes | its just when i ring phone 1 with my mobile and try and pick it up using phone 2 BLF |
| 14:18.29 | flackes | i just get a 603 error |
| 14:18.39 | *** join/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 14:19.04 | knhor | that sounds like you need to do "shared" line apperances |
| 14:19.25 | flackes | ok now im lost |
| 14:19.52 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found. |
| 14:19.52 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible... |
| 14:19.56 | flackes | that is the error |
| 14:20.02 | flackes | but it makes no sence to me |
| 14:20.24 | flackes | because if i call phone 1 with phone 2 and then get phone 3 to answer phone 2 with BLf it works |
| 14:20.34 | flackes | just wont work for calls comming from the Outside world |
| 14:21.58 | flackes | any ideas |
| 14:22.56 | knhor | when you hit the blf key, your dialing an extension. AFAIK you can't dial to ringing extension |
| 14:23.47 | flackes | well as far as i was aware if you press the BLF key when its no ringing it will dial that ext |
| 14:24.02 | flackes | but you should be able to press a flashing red BLF key to pick that call up |
| 14:24.13 | flackes | providing the SIP subscribes to the correct hint context |
| 14:25.03 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.20) |
| 14:25.18 | knhor | sorry, i've never messed with that, so I really can't speak to it. all i've ever done is blf as extension status and dial to extension. |
| 14:25.35 | flackes | doh |
| 14:25.38 | *** part/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 14:25.51 | flackes | what is this WIKI site? |
| 14:25.55 | mut | annyone ever worked with a lucent stinger, having a problem accessing the DS3-ATM profile |
| 14:26.10 | flackes | sorry mut i have no |
| 14:26.11 | flackes | not |
| 14:26.57 | knhor | voip-info.org ? - it is many/all things voip, include much (but not all) asterisk documentation |
| 14:27.38 | flackes | hmm been there, not seen anything about GXP-2000 and Asterisk |
| 14:27.49 | RoyK | knhor: it started off as an asterisk site, but now also holds some more |
| 14:28.21 | knhor | look carefully, it is there... i've used/seen it |
| 14:28.30 | flackes | is there a search thing there? |
| 14:28.32 | devel | yeah, like an entire page dedicated to it |
| 14:28.42 | flackes | omg why am i so blind then |
| 14:28.50 | flackes | maybe i have been looking at the wrong area |
| 14:28.51 | devel | probably been in IT too long |
| 14:29.05 | flackes | i have been looking for ASTERISK BLF |
| 14:30.00 | tzafrir | BLF=? |
| 14:30.05 | flackes | busy line filter |
| 14:30.09 | aydiosmio | BFD. |
| 14:30.17 | knhor | BusyLampField |
| 14:30.34 | flackes | lol every one calls it something different |
| 14:30.51 | brif8 | what causes "SIP/2.0 407 Proxy Authentication Required" (from tcpdump) I now have the CG 410 registering with * (can't have qualify = yes :( ) But when I dial 3000 3000 => DIAL(SIP/3000,20,Ttr) I get Address Incomplete on the snom 200 and this 407 in tcpdump ? |
| 14:30.55 | flackes | ok some one put me out my missery and tell me where the massive section is |
| 14:31.08 | brif8 | do I need an auth = |
| 14:31.22 | RoyK | would it be hard to allow only a certain amount of clients to login as one sip peer? as in "if already registered, refuse" and "if already registered, add new client with ip x.x.x.x" |
| 14:31.38 | knhor | as far is i know it is a Field (array) of Lamps that indicate Busy status.... but I didn't invent the term... so I'm probably wrong |
| 14:32.05 | aydiosmio | Proxy Auth Required is a authentication challenge in normal SIP communication |
| 14:32.09 | flackes | well i just get lost on the wiki site :P |
| 14:32.47 | aydiosmio | brif8: the 407 and your dialing problem are probably not related |
| 14:32.51 | knhor | goto the main page, find "ip phones" |
| 14:34.01 | flackes | ahh |
| 14:35.03 | *** join/#asterisk davidcsi (n=davidcsi@213.201.53.222) |
| 14:35.23 | flackes | dont see anything about BLF though |
| 14:35.49 | flackes | if anyone has found the page i need Please link :P |
| 14:35.59 | davidcsi | hello, O have two E1 spans working correctly but anyone knows why i get "chan_zap.c: Ring requested on unconfigured channel 0/31 span X"????? |
| 14:36.07 | knhor | goto the left and put BLF in the search box |
| 14:36.35 | Zouzou | how can i access Asterisk Web Voicemail afetr installing it(make webvmail)????? |
| 14:36.48 | flackes | davidcsi look at zaptel or zapta? both configered properly? |
| 14:37.28 | davidcsi | yes, both working fine |
| 14:37.29 | brif8 | aydiosmio: explain. When I dial I can watch the sip packets (SIP debug on) and the first thing the clipcomm does it report 407 Proxy ... |
| 14:37.31 | hank | am i correct when saying that FXO and FXS only apply to analog telephony and _not_ to isdn? |
| 14:37.41 | *** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net) |
| 14:38.07 | davidcsi | hank: yes |
| 14:38.20 | hank | YEEEHAA!!! thx :)) |
| 14:38.48 | davidcsi | flakes: i make and receive calls on both with no problem. but when the other side tries to send a call via timeslot 31 ansterisk reports that error. |
| 14:38.59 | aydiosmio | brif8: to authenticate with a challenge, client issues an INVITE, server sends back a 407 with a challenge, client responds with a hashed response, server then validates the credentials |
| 14:39.22 | aydiosmio | I think hank just won $10 |
| 14:39.48 | hank | aydiosmio: no that wouldnt have made me so happy :-p |
| 14:40.32 | davidcsi | anyone on the E1 error? |
| 14:40.35 | brif8 | aydiosmio: ok then what I need an auth = in the sip.conf ? |
| 14:41.48 | aydiosmio | brif8: if the phone isn't registering properly, yes |
| 14:42.09 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:42.12 | brif8 | aydiosmio: It shows it has in * console registered |
| 14:42.57 | flackes | knhor cant find anything about BLF and a GXP-2000 and ASterisk |
| 14:43.03 | flackes | its all for asterisk@home |
| 14:43.17 | brif8 | sip show peer 3000 has this MD5Secret : <Not set> Secret : <Set> |
| 14:43.25 | aydiosmio | brif8: does the client respond to the 407? |
| 14:43.58 | aydiosmio | hm, weird |
| 14:44.35 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 14:44.46 | Holos | Can anyone spot the problem in this Goto? GotoIf("SIP/100-b7a0d0a8", ""102" = "0"?notargetrn") It's still going to the notargetrn label. |
| 14:45.33 | brif8 | aydiosmio: http://pastebin.ca/191944 |
| 14:46.21 | aydiosmio | the call looks just fine |
| 14:46.37 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:46.41 | aydiosmio | it negotiated RTP, so it's not an authentication problem |
| 14:46.54 | *** part/#asterisk Zouzou (n=test@mail.splendor.net) |
| 14:47.03 | brif8 | ok then what ? |
| 14:47.56 | *** join/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
| 14:48.26 | aydiosmio | brif8: it's probably a dialing plan problem that I wouldn't be able to pinpoint |
| 14:48.44 | aydiosmio | run asterisk with full debugging to see what the digits failed |
| 14:48.47 | knhor | flackes: http://www.voip-info.org//tiki-pagehistory.php?page=GXP-2000&source=43 |
| 14:49.04 | slobberknocker | I am using moh-native and i have installed the asterisk-sounds, but i do not have a moh-native directory. can i just make a directory and put files in there? or does it have to be something special? |
| 14:49.05 | aydiosmio | easiest way is to start asterisk with -gdvvvvvvvv |
| 14:49.37 | brif8 | aydiosmio: all I have is 3000 => Dial(SIP/${EXTEN},20,Ttr) perhaps I need to somehow pass the Number it is to dial ? |
| 14:50.00 | aydiosmio | I can't help you beyond that |
| 14:50.32 | brif8 | ok thanks |
| 14:51.18 | *** join/#asterisk Ozii (n=Ozi@nat.office.legend.net.uk) |
| 14:53.20 | StiXantti | [TK]D-Fender: It seems, that the problem was SOMEHOW with our DNS server... |
| 14:54.11 | StiXantti | [TK]D-Fender: The "No NAT" in the log does not effect to my experience: Asterisk still send to WAN_IP:port |
| 14:55.10 | flackes | brif8 that will dial sip 3000 |
| 14:55.22 | flackes | Ttr are your timeouts and ringing |
| 14:56.29 | brif8 | yes How do I pass to 3000 that it is to dial another number ? something more like exten .X_ => DIAL(SIP/3000/${EXTEN},30,Ttr) would that work ? |
| 14:57.14 | flackes | well for example |
| 14:57.43 | flackes | you could do 1234 => Dial(SIP/4000,10) |
| 14:57.54 | flackes | that would mean if you dial 1234 it will ring sip 4000 for 10sec |
| 14:58.22 | flackes | ${EXTEN} = the number you enter in the phone so not always good to use this |
| 14:58.42 | flackes | exten => _7XXX,1,Ringing |
| 14:58.42 | flackes | exten => _7XXX,n,Wait(1) |
| 14:58.42 | flackes | exten => _7XXX,n,Answer() |
| 14:58.42 | flackes | exten => _7XXX,n,Set(FOO1=${CHANNEL:4}) |
| 14:58.42 | flackes | exten => _7XXX,n,Set(FOO2=${CUT(FOO1,-,1)}) |
| 14:58.43 | flackes | exten => _7XXX,n,Set(CALLERID(number)=${FOO2}) |
| 14:58.45 | flackes | exten => _7XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN}) |
| 14:58.49 | flackes | that what i use for all my internal ext |
| 14:59.01 | RoyK | ~pb |
| 14:59.03 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:59.12 | flackes | nerf its only 5 lines |
| 14:59.20 | flackes | <PROTECTED> |
| 14:59.29 | flackes | dont worrie about the setlines |
| 14:59.36 | RoyK | still some would call it flooding |
| 14:59.42 | flackes | yea supose |
| 14:59.51 | brif8 | right I realize that I can dial 1234 and it will dial SIP/3000 but how do I pass to SIP/3000 it is to dial a number ? |
| 15:00.10 | RoyK | flackes: are you trying to extract the sip id in that? |
| 15:00.14 | flackes | so you want to pass a call from 3000m to anouther one |
| 15:00.39 | flackes | RoyK i use that to set the caller ID of my internal lines |
| 15:00.50 | RoyK | ok |
| 15:00.53 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 15:00.54 | flackes | as i use the SIP caller id to set my exturnal caller id |
| 15:00.56 | aiksa[LV] | coppice: once again cleaned every file ic ould find of spandsp and old tiff libs on the machine. now i am ready to to do make install for asterisk |
| 15:01.31 | flackes | knhor that page did not seem to have anything about BLF just all about the phone.. am i blind |
| 15:01.51 | flackes | basicly im looking for someone that has BLF working with a GXP-2000 and asterisk |
| 15:02.33 | knhor | explain what your trying to acomplish again plz |
| 15:03.42 | brif8 | flackes: Yes I call SIP/3000 with a number 1234 and then SIP/3000 Dials 1234 (SIP/3000 is an FXO Gateway) |
| 15:04.19 | flackes | I have been trying to get my Grandstream busy line filter to work for ages.. |
| 15:04.19 | flackes | All the lights flash as they are supposed to. |
| 15:04.19 | flackes | If one Grandstream 7000 calls another Grandstream 7003 I can use Grandstream 7002 to pick the call up pressing the BLF button and all works fine. |
| 15:04.19 | flackes | However if I call Grandstream 7000 with a mobile phone and try to pickup the call with Grandstream 7002 all I get is a 603 error on Grandstream 7002. |
| 15:04.20 | flackes | I am using firmware 1.1.12 for the Grandstream and 1.2.12.1 version of asterisk |
| 15:05.37 | flackes | brif8 ok so i dial 1234 and i get sip/3000 then you want to pass that to your FXO line |
| 15:05.53 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:05.56 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 15:06.26 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found. |
| 15:06.27 | flackes | Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible... |
| 15:06.45 | flackes | and that is the error i get when i try and pickup the mobile phone call using BLF button |
| 15:07.13 | knhor | does the GS ring when you call from the cell phone? |
| 15:07.19 | flackes | yes |
| 15:07.26 | brif8 | flackes yes |
| 15:08.37 | knhor | so the inbound call does something like dial(sip/7000&sip/7001&sip/7002&sip/7003) ? |
| 15:08.49 | flackes | no no |
| 15:08.51 | flackes | ok |
| 15:08.58 | flackes | i call phone 7000 lets say |
| 15:09.01 | flackes | with my mobile |
| 15:09.22 | flackes | i then walk over to phone 7002 which has BLF on it watching 7000 |
| 15:09.39 | flackes | which ofc is flashing red because 7000 is rining |
| 15:09.56 | flackes | then when i press BLF on 7002 to pick up the call on 7000 i get a really nice 603 error |
| 15:10.09 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 15:10.29 | flackes | brif8 when sip3000 is ringing do you want to answer it then pass to the fxo or just get it to pass if the sip phone is not answered> |
| 15:10.30 | flackes | ? |
| 15:10.36 | eject_ck | how enable Instant messaging in Asterisk ? |
| 15:11.13 | knhor | flackes: I understand that you try to pickup with 7002, can you pickup with 7000 ? |
| 15:11.27 | flackes | yea if i answer the phone normaly |
| 15:11.46 | flackes | so if i pick 7000 up or use *8# on anouther phone it works fine |
| 15:12.11 | brif8 | flackes: pass it automatically |
| 15:12.41 | flackes | exten => s,1,Dial(${ARG2},20) |
| 15:12.41 | flackes | exten => s,2,Goto(s-4{DIALSTATUS},1) |
| 15:12.41 | flackes | exten => s-NOANSWER,1,Voicemail(u${ARG1}) |
| 15:12.41 | flackes | exten => s-NOANSWER,2,Goto(default,s,1) |
| 15:12.44 | *** join/#asterisk shodan (n=shodan@ip206.99-113-216.pppoe4.joliette.intermonde.net) |
| 15:12.46 | flackes | then use something like that tbh |
| 15:12.52 | flackes | i would put it in a macro |
| 15:13.02 | flackes | and i would make a global for ur fxo line |
| 15:13.39 | flackes | just change my voicemail to dia\ outbound fxo line ect |
| 15:13.58 | flackes | cant really give you any more because i dont use fxo |
| 15:14.12 | eject_ck | how enable support of IM ? |
| 15:14.21 | flackes | eject_ck as in msn? |
| 15:14.32 | flackes | for messages or for calls? |
| 15:14.34 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 15:14.40 | flackes | knhor any ideas? |
| 15:14.42 | flackes | lol |
| 15:14.49 | flackes | just wish someone has a full example |
| 15:14.56 | knhor | flackes: sorry, i'm out... :( |
| 15:15.00 | flackes | [TK]D-Fender you got any ideas |
| 15:15.06 | flackes | just dont know where to look now tbh |
| 15:15.25 | flackes | think i have something wrong with my code layout or something missing but i cant work out what |
| 15:15.43 | davidcsi | I got the solution, thank you guys. |
| 15:15.57 | flackes | davidcsi what was it? |
| 15:16.22 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-58-55.dsl.tul2ok.sbcglobal.net) |
| 15:16.34 | flackes | its silly i have 3 asterisk boxes all talking and a legacy siemens talking to the other boxes but cant get my BLF to work |
| 15:16.35 | flackes | lol |
| 15:16.45 | davidcsi | it was a problem with the zapata.conf, I had "channel => 48-61" |
| 15:16.57 | davidcsi | and it was "channel => 48-62" |
| 15:17.05 | flackes | STD E1 |
| 15:17.08 | eject_ck | flackes, EyeBeam have feature - send instant message to SIP peer - how enable it in asterisk ? |
| 15:18.02 | flackes | loadzone=uk |
| 15:18.03 | flackes | defaultzone=uk |
| 15:18.03 | flackes | span=1,1,0,ccs,hdb3,crc4 |
| 15:18.03 | flackes | bchan=1-15,17-31 |
| 15:18.03 | flackes | dchan=16 |
| 15:18.03 | flackes | span=2,2,0,ccs,hdb3,crc4 |
| 15:18.05 | flackes | bchan=32-46,48-62 |
| 15:18.07 | flackes | dchan=47 |
| 15:18.18 | flackes | found that is a nice layout |
| 15:18.27 | davidcsi | thats it |
| 15:18.41 | flackes | was the dchan that got me first |
| 15:18.43 | flackes | first |
| 15:18.58 | *** part/#asterisk knhor (n=knhor@cpe-70-125-158-177.satx.res.rr.com) |
| 15:19.11 | davidcsi | ok thnxs all |
| 15:19.41 | flackes | cya |
| 15:19.48 | davidcsi | bye |
| 15:19.55 | slobberknocker | looking at this, can anyone tell me why i am only able to display callerid name when someone calls 5202? Is it because i am running it through a macro? it displays google as the name and asterisk as the caller id number. i want it to show the callers number http://pastebin.ca/191976 |
| 15:20.33 | flackes | afk a sec will look in a mo |
| 15:20.35 | [TK]D-Fender | flackes: Sorry, I can't find any really good links, though I do recall seeing stuff about it around. Download the manual from Grandstream and look round. This may take some work. |
| 15:20.49 | flackes | already done that |
| 15:20.51 | [TK]D-Fender | eject_ck: * does not support SIP messaging even in passtrough, sorry. |
| 15:20.51 | flackes | nerf |
| 15:21.17 | flackes | this is what they sent back |
| 15:21.19 | flackes | Dear Andrew Shelton, |
| 15:21.19 | flackes | "603" means "Decline", which means Asterisk does NOT accept "**extention" to pick up the call in a pickup group. |
| 15:21.19 | flackes | Our reference is just for your reference ONLY. It will NOT work if you copy that configuration. We are not supporting Asterisk programing. |
| 15:21.19 | flackes | Your Asterisk programing is the source of the problem. Please google wiki or user forum to get some help or hints. |
| 15:21.21 | flackes | Thanks. |
| 15:21.45 | flackes | 1 i did not copy it :P and 2 what the hell is the point of it if it dont work :P |
| 15:21.51 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 15:22.08 | flackes | and now im here and still stuck :P |
| 15:23.02 | Ozii | Hello - question for the collective :) Using a TDM400P with FXO modules in the UK, when the remote user hangs up (they called us) we see the BT so called K-break i.e. disconnect supervision (100ms break), followed by 5s of tone, problem is it takes 7 seconds from the "k-break" until asterisk reports the zap channel as hungup - any ideas? |
| 15:23.28 | kratzers | AEL2 segfaults when an extension is followed by an empty block... is this an open issue, or should i report it? I don't see anything related in bugs.digium.com |
| 15:23.42 | brif8 | exten => 3000,Dial(SIP/3000/${EXTEN},20,Ttr) produces Invalid priority/label 'Dial' at line 353 Label missing trailing ')' at line 354 |
| 15:23.42 | brif8 | <PROTECTED> |
| 15:24.03 | [TK]D-Fender | brif8: Yeah... DUH, you have no priority on that line! |
| 15:24.43 | brif8 | DUHHH !!!!!!!!!!!! |
| 15:25.12 | [TK]D-Fender | brif8: exten => 3000,Dial....... <- BAD |
| 15:25.13 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-3afb443eee845822) |
| 15:25.20 | [TK]D-Fender | brif8: exten => 3000,1,Dial....... <- Good |
| 15:25.31 | [TK]D-Fender | (depending) |
| 15:25.47 | kratzers | like: context segfault { 1234 => { //will segfault } } |
| 15:26.03 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:26.08 | RoyK | kratzers: that's a feature! not a bug |
| 15:26.18 | xheliox | ~docs |
| 15:26.19 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:26.36 | flackes | brif8 i suggest you download Asterisk the future of telephony |
| 15:26.44 | kratzers | RoyK: asterisk dumps a core... wouldn't a warning be more suitable? |
| 15:26.53 | Corydon-w | kratzers: please report it on the bugtracker |
| 15:27.01 | RoyK | kratzers: i'm being ironic here...... |
| 15:27.07 | kratzers | will do |
| 15:27.11 | RoyK | kratzers: if asterisk crashes, report on bugs.digium.com |
| 15:27.20 | RoyK | 1.2 or 1.4? |
| 15:27.32 | flackes | so any one got any ideas or anything that can help me with my problem? |
| 15:27.37 | Corydon-w | RoyK: he said AEL2, so that's 1.4 or trunk |
| 15:27.45 | kratzers | RoyK: well, aelparse dies when doing aelparse -w |
| 15:27.47 | kratzers | trunk |
| 15:28.10 | kratzers | sorry, patched 1.2 |
| 15:28.42 | Corydon-w | You backported ael2 to 1.2, and wonder why it crashes? |
| 15:29.24 | RoyK | Corydon-w: did you get to look more into that bug of mine? #8087? |
| 15:29.25 | kratzers | no, I followed Murf's instructions on voip-info |
| 15:29.40 | kratzers | "NEW: For those of you running Asterisk 1.2, I've made a 1.2 based version of the AEL2 language." |
| 15:29.40 | Corydon-w | kratzers: report the bug, and murf will look at it |
| 15:29.44 | kratzers | will do |
| 15:29.54 | Corydon-w | RoyK: nope |
| 15:30.18 | brif8 | flackes: I d/l it from my bookshelf to my hands got a section in mind ? |
| 15:30.57 | flackes | Cool |
| 15:31.02 | flackes | sec |
| 15:31.21 | flackes | ok read from page 86 |
| 15:31.41 | flackes | that will tell you about the dial command |
| 15:31.52 | flackes | however i would suggest reading the whole thing a few times |
| 15:32.21 | flackes | it it worth emailing asterisk with this problem? |
| 15:33.27 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:33.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:33.46 | brif8 | ok Got it working at last :) :) :) now to go drink |
| 15:34.28 | Corydon-w | flackes: I read back, but I don't understand the problem |
| 15:34.32 | brif8 | what's a 1-stage FXO gateway ? |
| 15:34.43 | brif8 | vs a 2-stage FXO gateway ? |
| 15:34.47 | *** join/#asterisk saftsack (n=saftsack@p54A7FEAA.dip.t-dialin.net) |
| 15:35.16 | flackes | what that past place again :P |
| 15:35.30 | flackes | Corydon-w ill past you the whole email :D |
| 15:36.07 | Corydon-w | flackes: please use pastebin |
| 15:36.28 | *** join/#asterisk cfh (n=luca@82.193.23.3) |
| 15:36.42 | brif8 | pastebin.ca |
| 15:36.56 | brif8 | http://www.pastebin.ca/ |
| 15:37.04 | flackes | http://channels.debian.net/paste/3950. |
| 15:37.19 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:37.30 | flackes | Corydon-w basicaly when ever i try and pickup a call for anouther phone using my phones BLF i get a 603 error |
| 15:37.44 | Corydon-w | BLF? |
| 15:37.51 | flackes | busy lamp fild |
| 15:38.00 | flackes | field |
| 15:41.17 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-168-240.dsl.chcgil.sbcglobal.net) |
| 15:41.25 | Flauto | hi guys |
| 15:41.31 | Flauto | i have a question |
| 15:41.48 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 15:41.58 | flackes | Corydon-w what you think? |
| 15:42.03 | flackes | ever seen anything like this before? |
| 15:42.12 | Flauto | i am facing a problem that one of my remote sip user's port 5060 is blocked by isp |
| 15:42.13 | Corydon-w | flackes: try specifying the context it's actually dialling |
| 15:42.39 | Flauto | what can i do to make the user to be able to register a sip adapter to my asterisk server |
| 15:42.44 | Corydon-w | Flauto: that's between your user and his ISP |
| 15:43.03 | Flauto | can i define a different port for the user? |
| 15:43.10 | Flauto | or, there is nothing i can do |
| 15:43.54 | Corydon-w | Well, you could... but unless your user negotiates with his ISP, that's likely to get blocked, and you'll be right back where you started |
| 15:44.00 | flackes | Corydon-w ok.... but what context would i point it too? |
| 15:44.19 | Corydon-w | flackes: what context is it actually ringing? |
| 15:45.03 | flackes | well when BLF button is pressed it should exten => _**.,2,Pickup(${EXTEN:2}) |
| 15:45.24 | Flauto | well, this user is in china and chinese telecom is providing internet connections in china too, so they block anything possible to allow voip |
| 15:45.27 | Corydon-w | No, what context is ringing... not the call pickup |
| 15:45.35 | flackes | Stem |
| 15:45.43 | flackes | when the call comes in it would be stem |
| 15:45.51 | Corydon-w | so Pickup(${EXTEN:2}@stem) |
| 15:45.59 | Flauto | you mean that if i change it to a different port, they would block it again? |
| 15:46.04 | Flauto | then there is no hope man |
| 15:46.09 | flackes | ill give it a go |
| 15:46.21 | Flauto | blocking voip is their goal |
| 15:46.28 | flackes | wont stem then need something like exten =>7000,1,Dial(SIP/7000,20,r) |
| 15:46.29 | Corydon-w | Flauto: you could also set up a VPN between the endpoints |
| 15:46.46 | Flauto | hehe |
| 15:46.48 | Flauto | okay |
| 15:46.53 | Corydon-w | flackes: I haven't exhaustively examined your dialplan. |
| 15:47.04 | Flauto | i don't know vpn so i have to learn how to do it |
| 15:47.07 | hi365 | stupid question: asterisk -what to get the output in color? |
| 15:47.13 | flackes | well when the calls come in they go to stem and that dials the sip phone |
| 15:47.26 | Corydon-w | hi365: asterisk -vvvvvvvvvvvvvvvvc |
| 15:47.50 | flackes | then when the BLF button is pressed [BLF_group_pickup] |
| 15:47.54 | flackes | is accessed |
| 15:47.59 | hi365 | thanks. all thoes v's? |
| 15:48.03 | flackes | which should run exten => _**.,2,Pickup(${EXTEN:2}) |
| 15:48.16 | Corydon-w | hi365: not necessarily, but something like that |
| 15:48.34 | hi365 | not working here. |
| 15:48.34 | flackes | you dont have to put all the vss |
| 15:48.46 | flackes | you can just do asterisk -c |
| 15:48.47 | *** join/#asterisk cfh (n=luca@82.193.23.3) |
| 15:48.49 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 15:48.51 | flackes | then do set verbose 10 |
| 15:48.54 | hi365 | tahnsk |
| 15:49.05 | hi365 | but need verbose of 10? |
| 15:49.09 | Corydon-w | hi365: your terminal type must also support color |
| 15:49.38 | flackes | Corydon-w so pickup would need a context with the exten its going to pickup? |
| 15:49.46 | Corydon-w | hi365: and the name of the terminal in $TERM must be one that supports color |
| 15:49.58 | cfh | hi all, i m connect to a pri line with a server * and when i try to make a call with a SIPphone to a urban numer i dont heard the ring tone. |
| 15:50.02 | cfh | what can i do ? |
| 15:50.08 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:50.10 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:50.13 | Corydon-w | flackes: I don't use it, but try that |
| 15:50.29 | Corydon-w | flackes: it is mentioned in the application help |
| 15:51.57 | hi365 | Corydon-w; no color using putty... |
| 15:51.59 | Corydon-w | cfh: and you won't hear anything unless the PRI passes back a CALL PROCEEDING. That's the way it's supposed to work |
| 15:52.18 | Corydon-w | hi365: then you need to change your terminal type to something that supports color |
| 15:52.42 | hi365 | im no sure what u mean by "TemRinal type" |
| 15:52.51 | hi365 | im no sure what u mean by "Terminal type" |
| 15:53.03 | Corydon-w | "linux", "vt100", and "xterm-color" are all terminal types that support color |
| 15:53.20 | Corydon-w | hi365: type: echo $TERM |
| 15:53.37 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 15:53.42 | hi365 | xterm |
| 15:53.44 | *** join/#asterisk bmg505 (n=leon@c1-7-8.rndf.isadsl.co.za) |
| 15:53.56 | cfh | Corydon-w: but if i try a call to a mobile phone it works, wh ? |
| 15:53.59 | cfh | why ? |
| 15:54.16 | Corydon-w | cfh: that's a conversation you need to have with your PRI provider |
| 15:54.38 | flackes | Corydon-w same problem |
| 15:54.48 | flackes | just keep getting a 603 error |
| 15:55.12 | hi365 | Corydon-w: xterm |
| 15:55.31 | cfh | Corydon-w: what parameters can i change on my server voip ? |
| 15:55.54 | flackes | cfh you need a new client to access your server that supports colour |
| 15:56.02 | flackes | Eg penguie ect |
| 15:57.19 | Corydon-w | cfh: voip has nothing to do with it, as you've shown |
| 15:57.38 | Corydon-w | cfh: it's all about what the telco is or is not sending you |
| 15:57.41 | razu | anyone familiar with xorcom astribanks ? |
| 15:58.17 | *** join/#asterisk xezz (n=dddsd@b-h7.ektheseis.otenet.gr) |
| 15:58.24 | Corydon-w | hi365: look in your putty preferences. I think you'll find alternative terminal types in there |
| 16:00.33 | Corydon-w | flackes: oh, I see what the problem is |
| 16:01.32 | brif8 | I have the Clipcomm set a 1-Stage FXO Gateway for outbound calls from VoIP -> PSTN It rings which the clipcomm SIP/3000 answers and then goes silent while it dials the number given. even though I have Dial (SIP/3000/${EXTEN:2},20,r) the r parameter. any way to force the ring or something ? |
| 16:02.03 | Corydon-w | flackes: when you dial SIP/7003, the extension is actually 123454, not 7003 |
| 16:02.32 | hi365 | Corydon-w: i have a termianal type string. shal i change that? |
| 16:03.18 | Corydon-w | hi365: Please consult with the putty help. I'm not prepared to tell you what will work with that app |
| 16:03.24 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 16:03.41 | hi365 | no prob. tahnks |
| 16:06.19 | sudhir492 | D-Fender: In my sip.cfg, I have the following 2 lines: |
| 16:06.21 | sudhir492 | <alertInfo voIpProt.SIP.alertInfo.1.value="CUSTOM_1" voIpProt.SIP.alertInfo.1.class="8" |
| 16:06.21 | sudhir492 | <PROTECTED> |
| 16:06.25 | tzafrir | razu, I am |
| 16:06.36 | tzafrir | THough I'm about to go now |
| 16:07.05 | tzafrir | razu, check http://xorcom.com/drivers/astribank/zaptel-1.2.8.xpp.r2321/xpp/README.Astribank |
| 16:07.06 | aiksa[LV] | my zttest results are terrible |
| 16:07.06 | razu | tzafrir : what driver do I need to probe for this thing to work ... xpd_fxs ? |
| 16:07.17 | tzafrir | xpp_usb |
| 16:07.48 | razu | tzafrir : and in zaptel.conf i need the line fxoks=1-xx ? |
| 16:07.53 | aiksa[LV] | no wonder MeetMe had that lag |
| 16:08.05 | tzafrir | razu, right. |
| 16:08.07 | flackes | Corydon-w so what do i have to change |
| 16:08.24 | tzafrir | And again, check that doc (sorry, I don't have time right now) |
| 16:08.30 | Corydon-w | flackes: well, you could do it in a number of ways. |
| 16:08.39 | razu | tzafrir, ok thanks anyway :) |
| 16:08.40 | [TK]D-Fender | sudhir492: your Classes don't match. |
| 16:08.48 | Corydon-w | flackes: the easiest is probably to use a Goto instead of a Macro to dial the extension |
| 16:08.50 | flackes | Corydon-w any way that works is good |
| 16:08.54 | tzafrir | It's the one I've commited yesterday to the SVN |
| 16:09.04 | *** part/#asterisk cfh (n=luca@82.193.23.3) |
| 16:09.33 | tzafrir | razu, If you still have problems, I'll probably be back here as tzafrir_home or tzafrir_laptop in a few hours... |
| 16:09.48 | flackes | Corydon-w sorry what part needs changing? |
| 16:09.56 | sudhir492 | D-Fender: Ring Answer is class 8 later (in ringtype) |
| 16:10.01 | sudhir492 | What should I have |
| 16:10.06 | Corydon-w | flackes: so instead of the Macro, do a Goto(internal,7003,1) |
| 16:10.27 | sudhir492 | Should I leave the first one blank? |
| 16:10.37 | [TK]D-Fender | sudhir492: fine, show your "ring type" entries that should match for it... |
| 16:11.18 | sudhir492 | <PROTECTED> |
| 16:11.19 | sudhir492 | <PROTECTED> |
| 16:11.19 | sudhir492 | <PROTECTED> |
| 16:11.19 | sudhir492 | <PROTECTED> |
| 16:11.28 | sudhir492 | <PROTECTED> |
| 16:12.01 | flackes | Corydon-w why would that effect the pickup? |
| 16:12.11 | sudhir492 | D-Fender: Does that look good |
| 16:12.31 | Corydon-w | flackes: because then it's at an extension that IS 7003, instead of 123454 |
| 16:12.41 | sudhir492 | D-Fender: Do you want me use pastebin |
| 16:12.47 | Corydon-w | flackes: so the Pickup will find that extension |
| 16:13.17 | Corydon-w | flackes: the more complex way is to use a database lookup to map 7003 back to 123454 |
| 16:13.31 | flackes | ahh |
| 16:13.36 | flackes | like with call forwarding |
| 16:13.54 | Corydon-w | flackes: an easy way without changing anything would be to dial **123454 and see if it picks up |
| 16:13.54 | flackes | so store all the exturnal exten numbers with there internal ones |
| 16:14.27 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 16:14.56 | *** join/#asterisk clyrrad (n=ddd@TOROON01-1168097565.sdsl.bell.ca) |
| 16:15.33 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:18.31 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 16:19.43 | dhill | when a customer calls their number and is transferred to their voicemail system (not asterisk voicemail), their voicemail cannot hear the tones from the numbers pressed on the phone |
| 16:19.54 | dhill | would this be an asterisk setting.. or the end users sipura box? |
| 16:20.42 | Corydon-w | dhill: sounds likes they don't agree on the DTMF settings |
| 16:20.51 | dhill | right |
| 16:20.54 | *** join/#asterisk pingwin (i=pingwin@gateway/tor/x-c9e9ac4e960b6da3) |
| 16:20.59 | Corydon-w | So make them agree |
| 16:21.06 | dhill | i am guessing i need to change some settings on their sipura |
| 16:21.23 | flackes | Corydon-w OMG thanks very much.... dont know why i did not see that |
| 16:21.33 | Corydon-w | At the very least, you need to know what the settings are on their device |
| 16:21.48 | flackes | Corydon-w Taken me about 2 months to get that working |
| 16:22.10 | dhill | Corydon-w: such as dtmf playback level, playback length? |
| 16:22.14 | pingwin | i have a big problem. My asterisk system is remembering the actions that occur when someone calls in. So if it's going to a dial function that called multiple phones, it keeps routing the number when it calls back to the extention that picked it up prior. any idea what could be doing this? |
| 16:22.38 | pingwin | ideally I'd like to prevent asterisk from "remembering" the actions of any call minus the cdr |
| 16:22.38 | Corydon-w | dhill: no, dtmfmode=info or rfc2833 |
| 16:22.45 | brif8 | http://pastebin.ca/192030 is what the console shows when the call is made ? |
| 16:22.45 | dhill | ok |
| 16:22.56 | flackes | cya tomorrow peeprs time to go home |
| 16:23.08 | flackes | thanks again for all ur help :D |
| 16:23.16 | p1p | Anyone around that uses polycom phones with the latest firmware? |
| 16:24.08 | dhill | ok, thanks |
| 16:25.26 | *** join/#asterisk eIIisdee (n=ellisdee@69.15.174.114) |
| 16:25.56 | xheliox | Does anyone know what the max length of a caller ID name string can be? |
| 16:26.30 | pingwin | i have a big problem. My asterisk system is remembering the actions that occur when someone calls in. So if it's going to a dial function that called multiple phones, it keeps routing the number when it calls back to the extention that picked it up prior. any idea what could be doing this? |
| 16:26.38 | xheliox | (that will be sent via PRI) |
| 16:26.41 | *** join/#asterisk ast_freak (n=jesse@h69-130-173-212.69-130.unk.tds.net) |
| 16:27.06 | eIIisdee | i have a question: how do i remotely restart phones from the asterisk cli |
| 16:27.09 | pingwin | not sure what max is, how long of a string are you trying to send? |
| 16:27.15 | eIIisdee | i am trying to reboot polycom telephones |
| 16:27.24 | eIIisdee | sip notify polycom <peer>? |
| 16:27.58 | xheliox | pingwin: Long story. Just trying to prove to a sales engineer at Embarq that he's full of shit. |
| 16:28.32 | xheliox | pingwin; He wants me to send a full address via the CNAME for 911 calls and he claims that will get passed to the PSAP.. and like I said, I dont think so.. for a number of reasons. |
| 16:28.40 | *** join/#asterisk neonluc (i=luc2@modemcable046.12-80-70.mc.videotron.ca) |
| 16:28.53 | pingwin | yeah I don't think so either |
| 16:28.56 | pingwin | are you in the states? |
| 16:29.07 | xheliox | Yeah. |
| 16:29.29 | *** join/#asterisk elduffy (n=elduffy@pd9569322.dip0.t-ipconnect.de) |
| 16:29.58 | pingwin | yeah then I don't think you'd get much use out of doing that. wouldn't think cid would be able to contain that much data |
| 16:30.12 | neonluc | hello jaimerais to have the guide to install your software |
| 16:31.01 | elduffy | hello, anybody a clue on zaprtc? |
| 16:32.04 | neonluc | Asterisk Version 1.2.12.1 |
| 16:32.16 | sudhir492 | D-Fender: Will you please take a look at the following - http://pastebin.ca/192036 I have given all the relevant details there. What is wrong there? |
| 16:32.51 | xheliox | pingwin: *nod* I agree. That's why I'm trying to get facts to shove down this guy's throat. :) |
| 16:32.59 | neonluc | which which is better for linux red hardware 9.0 |
| 16:34.21 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 16:34.24 | sudhir492 | eIIisdee: You are right, you can do something like: sip notify polycom-check-cfg 412 |
| 16:35.25 | eIIisdee | aah, i see now. |
| 16:35.35 | eIIisdee | i checked out /etc/asterisk/sip_notify.cfg |
| 16:35.47 | eIIisdee | i guess i just append manufacturer name along with the event name. |
| 16:38.04 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:38.33 | eIIisdee | must debug on in order to see the results of a notify polycom check cfg? |
| 16:38.56 | elduffy | issue w/ suse and zaprtc not willng to load... anybody a clue? |
| 16:40.20 | *** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 16:42.48 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 16:42.55 | key2 | !seen zoa |
| 16:43.06 | key2 | ~seen zoa |
| 16:43.12 | jbot | zoa <n=d@pirus.securax.be> was last seen on IRC in channel #asterisk, 34d 4h 28m 28s ago, saying: 'and on an E1 they usually dont to my experience'. |
| 16:44.06 | hardwire | ~fishslap key2 |
| 16:44.09 | jbot | ACTION slaps key2 up side the head with a wet fish. |
| 16:44.09 | hardwire | aww |
| 16:44.14 | hardwire | yay! |
| 16:45.09 | neonluc | is what is necessary that junstall all the software one which has on your site or right to have Asterisk Version 1.2.12.1 is coraite |
| 16:51.14 | [TK]D-Fender | sudhir492: "s,3" looks bad, can't imagine its purpose. Your "s,2" line should be SIPAddHeader(Alert-Info: Ring Answer) |
| 16:52.38 | [TK]D-Fender | neonluc: Demande en fracais, on va mieux vous comprendre :) |
| 16:54.25 | neonluc | je me demande si il faut prendre plus qun loficielle pour que ca marche |
| 16:54.38 | neonluc | il me faut tu juste Asterisk Version 1.2.12.1 ou les autre |
| 16:55.00 | neonluc | et coment bien le configurée |
| 16:56.27 | aptura | Who here has a asterlink account? |
| 16:56.45 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 16:56.51 | *** join/#asterisk elduffy (n=elduffy@pd9569322.dip0.t-ipconnect.de) |
| 16:58.36 | [TK]D-Fender | neonluc: Ca depends sur les fonctionalites que tu as besoins d'avoir. |
| 16:58.52 | [TK]D-Fender | neonluc: Que fereais-vous avec? |
| 16:59.12 | marcus2 | hrm, still no luck finding a reliable voip toll-free provier :/ |
| 17:00.27 | syzygyBSD | marcus2: let me check what a couple clients are using |
| 17:00.57 | aptura | syngyBSD also let me know to. |
| 17:01.10 | *** part/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 17:01.20 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-168-51.bflony.east.verizon.net) |
| 17:01.31 | SuPrSluG | hello al |
| 17:01.36 | SuPrSluG | all |
| 17:01.38 | aptura | asterlink main number picks up and gives the IVR extention anouncment but thay are dead. |
| 17:01.50 | neonluc | moi c est plus pour téléphoner et surement en vendre pour palrler et surement pour les londistence aussi |
| 17:02.16 | aptura | neoloc google bablefish |
| 17:03.42 | syzygyBSD | hmm, interesting, this client appears to just have the 800 number forwarded to a local did over zap.... |
| 17:04.35 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 17:05.11 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:05.22 | aptura | neoloc, Pouvez-vous parler anglais ? Vous pouvez employer ce site Web au Français de translace en anglais situé à http://babelfish.altavista.com/tr. |
| 17:05.39 | *** join/#asterisk adorah (n=admin@87.68.169.166.cable.012.net.il) |
| 17:05.44 | syzygyBSD | lol, nice |
| 17:06.17 | aptura | syzygyBSD which 1800 carriers do you know of that support asterisk? |
| 17:06.35 | syzygyBSD | looking at other clients now... |
| 17:06.48 | [TK]D-Fender | neonluc: Qu'est-ce que tu veut utiliser comme telephone? Et coome veut-tu te rendre au reseau telephonique regulier? |
| 17:06.48 | syzygyBSD | phone.gsihosting.com is what one uses, but that is their host too i believe |
| 17:07.01 | aptura | how reliable are thay? |
| 17:07.01 | neonluc | me C is more to telephone and surely to sell some for palrler and surely for the londistence too |
| 17:07.19 | syzygyBSD | I haven't heard any complaints by the client about them |
| 17:07.41 | syzygyBSD | but the 800 goes into an automated only service... so they really wouldn't hear any |
| 17:08.10 | nicchap | has anyone out there used speechBackground (lumenvox) succesfully. The rec part works, but the file doesn't play. This starts happening after a few calls to it. |
| 17:08.20 | syzygyBSD | and you know you can trust a company that has the default centos apache page on their site |
| 17:08.24 | aptura | neoluc, Pouvez-vous reformuler cela ? Il ne circule pas bien sur l'écran. |
| 17:08.32 | hi365 | ~ |
| 17:08.42 | jmls | anyone know of a internet order muffin service that delivers in the US ? |
| 17:09.05 | aptura | syzygyBSD yea I am having a issue finding a toll free 1800 service also. |
| 17:09.37 | [TK]D-Fender | Hey, OT question I need a hand with : Can someone help me make a 1-line IPTABLES rule to filter out in incoming port? |
| 17:10.00 | [TK]D-Fender | I have a fixed IP on the dest addr inbound and the devname is "w1g1" |
| 17:10.05 | *** join/#asterisk Yogik (n=Miranda@c-66-41-255-50.hsd1.mn.comcast.net) |
| 17:10.11 | [TK]D-Fender | Should be dead simple and I need it fixed up damn fast.... |
| 17:10.18 | syzygyBSD | jmls: what kind of 'muffins' are we talking about here |
| 17:10.39 | syzygyBSD | [TK]D-Fender: the rest of iptables setup right? |
| 17:10.42 | aptura | syzygyBSD I think he can google that info. |
| 17:11.01 | [TK]D-Fender | syzygyBSD: Yeah, basic 4 line setup for NAT, and thats it. |
| 17:11.04 | syzygyBSD | heh, ya, but some of the conversations we have had here I dont' feel bad at all answering him |
| 17:11.36 | [TK]D-Fender | aptura: Yeah, I'm jsut not finding a good example for this 1 line job.... |
| 17:11.36 | [TK]D-Fender | but I am still looking at least. |
| 17:11.36 | eIIisdee | anyone have issues with polycom phones not updating time/date correctcly? |
| 17:11.36 | [TK]D-Fender | this is a 10 second answer from anyone with experience |
| 17:11.49 | eIIisdee | i have ntpd configured correct on the asterisk box.. hwclock displays the correct time. |
| 17:11.59 | jmls | syzygyBSD: the muffins the US guys seems to like. There's a great site for the uk: http://www.bakinboys.co.uk |
| 17:12.20 | eIIisdee | in sip.cfg i have the appropriate credentials for the sntp server. |
| 17:13.16 | elduffy | is anybody familiar with ZAPRTC? |
| 17:13.33 | Yogik | eIIisdee , do ntpq -pn and see if your server is synced with upstream servers , if not - it will not give time to clients |
| 17:14.24 | syzygyBSD | iptables -A chain reject --source-ports 22 |
| 17:15.02 | SuPrSluG | I am having a problem with hearing the ring when placing a call. When I call the number it answers and goes to my IVR. When I select option 1 id dials and says its ringing that number. But, I hear nothing until it goes to voicemail.. Any ideas? |
| 17:15.19 | *** join/#asterisk eBody (n=icechat5@207.71.51.162) |
| 17:15.32 | Yogik | DTMF settings are wrong |
| 17:15.36 | syzygyBSD | SuPrSluG: can you still pick up the line while it is ringing? |
| 17:15.51 | eBody | is there a channel for asterFax support? |
| 17:16.00 | myiagy | [TK]D-Fender what exactly do you want to do? block access to a certain port? |
| 17:16.51 | eIIisdee | Yogik, server is syncing. i have evidence of this via my messages log. |
| 17:17.25 | eIIisdee | if it helps my phones have constant red lights on. i give.. the phones are retrieving ip addresses via dhcp. |
| 17:18.10 | eIIisdee | =P |
| 17:18.10 | SuPrSluG | syzygyBSD:I'm doing it remotely. Setting up an 800 number. It appears to be working properly, just no ringing tone play to the caller. |
| 17:18.10 | syzygyBSD | oh.. I hate the security someone setup on this server 4 years ago |
| 17:18.10 | syzygyBSD | I can only update the time 1 second at a time |
| 17:18.18 | syzygyBSD | since the clock is 20 minutes off, it will take 20 minutes to update to the correct time |
| 17:18.25 | neonluc | moi c est surtout pour téléphoner mes amis avec et pour faire des londistence aussi si posible mes londiatence pas importemp ou que ca marche aumoin entre nous |
| 17:19.30 | syzygyBSD | SuPrSluG: eh, just add a r option to the dial() command |
| 17:20.14 | SuPrSluG | i'll try that. Didn't have to do that for zap channels. |
| 17:20.31 | aiksa[LV] | has anyone seen this error before and know how to deal with that: Don't know what to do with control frame 15 |
| 17:21.36 | nicchap | aiksa: Seen it on PRI calls, I just neglect it. |
| 17:22.55 | SuPrSluG | that didn't work. but the person answered the phone so it is ringing. |
| 17:23.08 | aiksa[LV] | nicchap: well i am trying to send out fax with txfax |
| 17:23.30 | aiksa[LV] | this is the last message before the process stops and page doesnt get sent |
| 17:23.48 | aptura | SuPrSluG are you using a 1800 service? |
| 17:23.55 | SuPrSluG | nufone |
| 17:24.25 | aptura | good luck! lack of customer service sucks ;) |
| 17:24.38 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 17:24.57 | [TK]D-Fender | myiagy: Yes. |
| 17:25.25 | myiagy | [TK]D-Fender iptables -A INPUT -p tcp --dport portnum -j DROP |
| 17:25.26 | SuPrSluG | yeah. once ya get em working they're reliable, BUT a problems can go unattended. |
| 17:25.41 | [TK]D-Fender | myiagy: Great, thanks |
| 17:25.45 | myiagy | np |
| 17:32.21 | *** join/#asterisk DonX (i=don@the.lostserver.net) |
| 17:32.52 | DonX | Hi all...I have a question about GotoIf |
| 17:32.59 | DonX | I'm currently doing this... |
| 17:33.14 | DonX | exten => _6XXXXX,1,Goto(site${EXTEN:1:1},${EXTEN:1},1) |
| 17:33.42 | DonX | can I change that to a gotoif the site# exists as a context? |
| 17:34.09 | DonX | otherwise I would like it to play a recording saying the number is unavailable |
| 17:34.57 | [TK]D-Fender | DonX: if it doesn't then you dialplan will continue on to Priority 2. Thats where. |
| 17:35.22 | DonX | oh ok, so I can just put in a recording there? |
| 17:35.26 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:35.40 | Bobcat991966 | Hello All, does anybody have an idea why my terminal screen keeps repeating Asterisk died with code 1 but asterisk seems to be running? |
| 17:35.41 | DonX | like...playback(unavailable) ? |
| 17:36.08 | jmls | Bobcat991966: are you using safe_asterisk ? |
| 17:36.21 | Bobcat991966 | I believe so. |
| 17:36.40 | jmls | Bobcat991966: safe_asterisk automatically restarts asterisk |
| 17:36.46 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 17:37.18 | Bobcat991966 | That make sence. I read that somewhere but how can it keep dying and restarting without some of my calls failing? |
| 17:37.28 | [TK]D-Fender | DonX: Yup |
| 17:37.30 | heison | service button on my 7960 pointing at my webserver works fine until I upgraded from SIP 7.5 to SIP 8.4, now I get BTXML Error when I press the service button, and I can't seem to find documents on google regarding this particular problem. |
| 17:37.53 | jmls | ah! you are starting safe_asterisk when asterisk *is already running* |
| 17:38.17 | jmls | so it can't restart :) |
| 17:38.17 | Bobcat991966 | how do I previent that jmls? |
| 17:38.18 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 17:38.33 | Bobcat991966 | that makes even more sence. |
| 17:38.55 | Bobcat991966 | Im not a linux guru can you help me previent that? |
| 17:39.13 | jmls | do you have asterisk automatically starting when you boot up ? You may be trying to start it twice .. |
| 17:39.19 | Bobcat991966 | ye |
| 17:39.22 | Bobcat991966 | yes |
| 17:39.53 | jmls | cd /etc/rc.d |
| 17:40.02 | Bobcat991966 | Let me look |
| 17:40.04 | jmls | find . -name *asterisk* -print |
| 17:40.20 | trevarthan | Can someone tell me if a 200mhz gumstix is fast enough to run a SIP phone like linphone or twinkle with ulaw? |
| 17:40.38 | jmls | trevarthan: someone can |
| 17:40.44 | jmls | sorry, couldn't resist |
| 17:41.29 | *** part/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 17:41.30 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 17:41.54 | trevarthan | does sip support stereo audio? or is it just mono? |
| 17:43.40 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:43.56 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:45.08 | trevarthan | hmmmm... I'm guessing not. |
| 17:45.21 | trevarthan | darn. I was hoping to run 3d audio processing on a sip channel |
| 17:46.03 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:46.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:49.17 | pingwin | i have a big problem. My asterisk system is remembering the actions that occur when someone calls in. So if it's going to a dial function that called multiple phones, it keeps routing the number when it calls back to the extention that picked it up prior. any idea what could be doing this? |
| 17:49.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:49.44 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.31) |
| 17:49.47 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 17:50.18 | jmls | pingwin: that's got to be down to your dialplan |
| 17:50.22 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 17:50.39 | jmls | asterisk does not "remember" anything unless the diaplan saves "state" |
| 17:50.44 | jmls | *dialplan |
| 17:51.05 | pingwin | jmls: k, what kind of macro/function would be called to save the state? |
| 17:52.53 | jmls | pingwin: dbget/dbput |
| 17:53.12 | pingwin | yeah, that's not being used at all |
| 17:53.18 | benjk | there is no way to save the state, its already screwed, no chance, dead and buried |
| 17:53.26 | aiksa[LV] | perhaps someone would know where txfax puts it log, if it has debug enabled in application comand? |
| 17:53.33 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:53.34 | Bobcat991966 | jmls: I do not have a file called rc.d, I do have a directory though |
| 17:53.37 | jmls | benjk: you live in the UK then ? |
| 17:53.43 | benjk | heh |
| 17:53.53 | jmls | Bobcat991966: that's right. it is a directory |
| 17:53.57 | jmls | cd /etc/rc.d |
| 17:53.58 | benjk | no, but doesn't it apply to any arbitrary state? |
| 17:54.16 | [TK]D-Fender | pingwin: Show us CLI output of this "repeat" call, and the dialplan that does it. |
| 17:54.18 | jmls | we've got the Blair witch project at the helm... |
| 17:54.50 | pingwin | [TK]D-Fender: how verbose? |
| 17:54.59 | eIIisdee | damn |
| 17:55.01 | benjk | and we've got a little Hitler as next door neighbour |
| 17:55.11 | eIIisdee | figured out why phones werent updating |
| 17:55.22 | jmls | benjk: which one ? |
| 17:55.25 | benjk | a Korean one |
| 17:55.30 | jmls | ah. |
| 17:55.33 | hardwire | anybody here worked with inter-tel phone systems? |
| 17:55.47 | benjk | he's kidnapping folks from his U-boats off our beaches |
| 17:55.48 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:55.50 | hardwire | and if so.. how did you control the urge to huck it out the window at a passing train. |
| 17:56.00 | benjk | he's firing missiles over our heads |
| 17:56.01 | eIIisdee | something as simple as a missng < |
| 17:56.03 | jmls | benjk: make sure that your * servers are emp protected :) |
| 17:56.04 | sudhir492 | D-Fender: Will you please take a look at this - http://pastebin.ca/192036 |
| 17:56.07 | Bobcat991966 | what file in the directory should I be looking at...there are several. There is a file called rc, one call rc local and one called rc.sysinet no of which have the phrase -name *asterisk* -print. |
| 17:56.09 | benjk | and now he wants to test a nuke |
| 17:56.21 | eIIisdee | in my sip.cfg |
| 17:56.32 | jmls | Bobcat991966: just cd /etc/rc.d |
| 17:56.37 | jmls | then type the command |
| 17:56.43 | jmls | find . -name *asterisk* -print |
| 17:56.48 | jmls | do you get anything from that ? |
| 17:56.49 | benjk | but those stupid American idiots think that they better wasted their attention on this guy in Iraq |
| 17:56.55 | sudhir492 | D-Fender: sorry, I missed your remark. |
| 17:57.00 | sudhir492 | Thanks |
| 17:57.33 | marcus2 | so i got my second linksys wrt/asterisk server up and running |
| 17:57.37 | [TK]D-Fender | pingwin: "10" |
| 17:57.42 | marcus2 | works really well hosting the polycom 601 |
| 17:57.49 | [TK]D-Fender | pingwin: use www.pastebi.ca please |
| 17:57.54 | benjk | North Korean diplomats are also the biggest drug traffickers on the planet |
| 17:57.54 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:57.55 | [TK]D-Fender | pingwin: use www.pastebin.ca please |
| 17:57.56 | syzygyBSD | what is the easiest way to get a 30 second of silence recording? |
| 17:58.18 | syzygyBSD | the silence directory only has up to 10 |
| 17:58.19 | jmls | ask tony blair for his future plans |
| 17:58.25 | [TK]D-Fender | syzygyBSD: there is a slient recording in the sounds folder already |
| 17:58.28 | caio1982 | tzafrir: are you noticying some problem with the alioth mailer? i cannot send messages to the pkg-voip list , theyre coming back with a local delivery failure at haydn |
| 17:58.31 | eIIisdee | record with no mic plugged in |
| 17:58.31 | Bobcat991966 | Ok now i understand jmls. this is the output |
| 17:58.38 | Bobcat991966 | ./rc6.d/K60asterisk |
| 17:58.40 | Bobcat991966 | ./init.d/asterisk |
| 17:58.41 | Bobcat991966 | ./rc4.d/S40asterisk |
| 17:58.42 | [TK]D-Fender | syzygyBSD: Just do it 3 times |
| 17:58.43 | Bobcat991966 | ./rc1.d/K60asterisk |
| 17:58.44 | Bobcat991966 | ./rc0.d/K60asterisk |
| 17:58.46 | Bobcat991966 | ./rc5.d/S40asterisk |
| 17:58.48 | Bobcat991966 | ./rc3.d/S40asterisk |
| 17:58.49 | Bobcat991966 | ./rc2.d/S40asterisk |
| 17:58.56 | jmls | yikes |
| 17:58.57 | syzygyBSD | [TK]D-Fender: but that would be too easy... |
| 17:59.04 | syzygyBSD | damn, now i feel stupid |
| 17:59.12 | benjk | so you have this totally out of whack running amok weirdo country and the Americans stick their heads in the sand while making a lot of fuzz about other places that aren't even beginning to smell one bit as dangerous |
| 17:59.12 | syzygyBSD | thanks a lot |
| 17:59.13 | [TK]D-Fender | syzygyBSD: Thereby making it the easiest :) |
| 17:59.17 | jmls | i've just got init.d asterisk |
| 17:59.35 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 17:59.36 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 17:59.43 | jmls | rene1 !! |
| 17:59.51 | Bobcat991966 | hmmm, I wonder ehy the differnce> |
| 17:59.55 | rene1 | jmls: hey man |
| 18:00.03 | rene1 | your sugestion worked very well for me! |
| 18:00.30 | rene1 | using "/n" is what i needed |
| 18:00.41 | jmls | rene1: file fixed the /n transfer problem. He hasn't yet submitted it as a patch just yet, but it works for me ! |
| 18:00.42 | rene1 | i wonder what does it does exactly |
| 18:00.48 | rene1 | really |
| 18:00.51 | rene1 | excellent |
| 18:01.09 | aydiosmio | voodoo magic |
| 18:01.10 | jmls | "/n" means keeps the channel in the loop, don't destroy it |
| 18:01.25 | jmls | "/n" "no destroy" |
| 18:01.31 | jmls | or something like that. |
| 18:01.35 | rene1 | the only thing i dislike about local channels is all the Rename stuff that shows up in AMI |
| 18:02.26 | rene1 | is there a way to get the "real channel" for a local channel? |
| 18:02.35 | rene1 | from the dial plan? |
| 18:02.44 | Bobcat991966 | I wonder if I should delete all the KXXasterisk and just leave the init.d/asterisk |
| 18:03.00 | jmls | I set an __ variable before entering the local channel |
| 18:03.04 | jmls | e.g. |
| 18:03.15 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:03.19 | jmls | Set(__QUEUECHANNEL=${CHANNEL}) |
| 18:03.26 | jmls | dial(local/something/n) |
| 18:03.33 | jmls | then in local/something |
| 18:03.49 | jmls | ${QUEUECHANNEL} has the original channel |
| 18:03.57 | rene1 | i see.. but say you are originating to a local channel |
| 18:04.09 | rene1 | and then the local channel does a dial(zap/g1/exten) |
| 18:04.23 | rene1 | you dont know the channel g1 is going to be do you? |
| 18:04.41 | jmls | I was tlaking about inbound |
| 18:04.44 | jmls | *talking |
| 18:04.54 | jmls | why do you want to use for outbound ? |
| 18:06.50 | rene1 | mm say i have a sip - zap call, are there any differences if i hangup the sip or zap leg? i mean from the dialplan perspective? |
| 18:07.03 | jmls | use the g option |
| 18:07.06 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 18:07.09 | jmls | on the dial |
| 18:07.11 | rene1 | g? |
| 18:07.36 | jmls | exten => _X.,n,Dial(Zap/G3/${EXTEN},120,g) |
| 18:07.54 | jmls | g allows the dialplan to continue after the call has ended |
| 18:08.02 | rene1 | i see |
| 18:08.10 | jmls | h extension catches the other hangup |
| 18:08.13 | rene1 | i can get the channel there |
| 18:08.16 | Inverted | is there a way in the dialplan to call a script which can authenticate a user stored in a database on a remote machine? |
| 18:08.21 | rene1 | cool |
| 18:08.34 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.155.94) |
| 18:08.41 | rene1 | Inverted AGI or the mysql dialplan command |
| 18:08.44 | jmls | rene1: if you want to get the channel during the call, use the M option and run a macro when the call is connected |
| 18:09.13 | jmls | exten => _X.,n,Dial(Zap/G3/${EXTEN},120,gM(connected^${JabberID}^${AgentID}^${EXTEN}^${UNIQUEID}^Peer=${DIALEDPEERNAME})) |
| 18:09.33 | *** join/#asterisk Holos (n=asdf@204.101.26.106) |
| 18:10.10 | Holos | I'm getting stuck with GotoIf.. exten => _s,n,GotoIf(1 = 2?true:false) That should go to false, but it goes to true. |
| 18:10.12 | syzygyBSD | what does "Don't know what to do with control frame 15" mean? |
| 18:10.40 | rene1 | jmls: great |
| 18:10.46 | jmls | syzygyBSD: we're all going to die |
| 18:10.50 | Corydon-w | Holos: GotoIf($[1 = 2]?true:false) |
| 18:10.50 | rene1 | i do need to get the zap channel |
| 18:10.55 | rene1 | dring the call |
| 18:11.01 | syzygyBSD | well, I'm goign to go get drunk then |
| 18:11.06 | jmls | :) |
| 18:11.15 | Holos | Corydon-w: Thanks.. I'll try that. |
| 18:11.15 | rene1 | i also used to get that with 1.2.x a lot |
| 18:11.43 | jmls | rene1: a great way of seeing what you have as variables is to use the dumpchan command in the dialplan - this spits out all kinds of info |
| 18:12.38 | rene1 | i have used it inside the local channels, but have never ever been able to get the original zap channel info.. i am getting it tru a dirty AMI status hack but your option is way cooler |
| 18:13.16 | rene1 | via ami status action, and looking for dialed number.. that is |
| 18:14.15 | rene1 | jmls: would i be able to read ${CHANNEL} inside the macro? |
| 18:14.53 | rene1 | seen ~oej |
| 18:16.07 | MikeJ | ~seen oej |
| 18:16.19 | jbot | oej <n=oej@23.Red-88-7-53.staticIP.rima-tde.net> was last seen on IRC in channel #asterisk, 7d 3h 3m 47s ago, saying: '~seen kpfleming'. |
| 18:16.21 | rene1 | thanks |
| 18:16.36 | MikeJ | hehe.. |
| 18:16.50 | file | MikeJ: I see you! |
| 18:16.59 | MikeJ | not ture |
| 18:18.37 | *** join/#asterisk Bpedersen (n=bp@82-192-171-74.sk.dsl.struer.net) |
| 18:18.49 | hmmhesays | does perl have a switch statement? |
| 18:19.41 | file | MikeJ: you are... right there! |
| 18:20.21 | MikeJ | so.. what do you say.. asterisk_politics mailing list? |
| 18:20.32 | hmmhesays | heh |
| 18:20.43 | file | MikeJ: sounds political |
| 18:22.05 | MikeJ | see asterisk-biz mailing list lately? |
| 18:22.47 | syzygyBSD | so is "channel.c:2483 __ast_request_and_dial: Don't know what to do with control frame 15" something that will always happen when dialing from a local channel? |
| 18:22.54 | MikeJ | well.. I registered #asterisk-politics irc chan just to be sure ...:P |
| 18:23.16 | rene1 | syzygyBSD: i saw that without using local channels |
| 18:23.26 | rene1 | when originating via ami |
| 18:23.31 | MikeJ | and #asterisk-religion but I've had that sucker for a long time.. |
| 18:23.35 | rene1 | with 1.2.x |
| 18:23.48 | syzygyBSD | well, I know it will happen other times, but it seems to happen consistantly when using local channels |
| 18:24.21 | rene1 | i happened every time for me. now it is gone (svn trunk) |
| 18:24.43 | rene1 | and now i am using local channels |
| 18:26.31 | *** join/#asterisk E-Rage (n=erage@nat-gw.seattle.identitext.com) |
| 18:26.33 | jmls | rene1: yes, ${CHANNEL} is the zap channel in the macro |
| 18:27.18 | rene1 | cool |
| 18:28.18 | *** join/#asterisk cp5 (n=cp5@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 18:30.05 | E-Rage | I've got an interesting one: is there a way to pass PRI d-channel call parameters when you're bridging Zap channels in a call? |
| 18:30.40 | E-Rage | What I think is happening is a legacy PBX is defing a call as international (a la pridialplan in zapata.conf) per call, stripping the 011 that was dialing, and passing along |
| 18:30.48 | *** part/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 18:31.21 | E-Rage | However, passing just that to the PSTN doesn't work, because the call lacks an 011, and is not defined as internaltional and thus the number is invalid |
| 18:31.43 | E-Rage | So my thinking is that there ought be a way to pass that along in a call |
| 18:31.45 | E-Rage | Any thoughts? |
| 18:32.50 | sudhir492 | D-Fender: I changed the extensions.conf as you suggested, still the Polycom phone keeps on ringing and ringing |
| 18:33.00 | sudhir492 | http://pastebin.ca/192140 |
| 18:36.16 | jmls | E-Rage: if the legacy pbx is only passing the number without the 011, then surely the exchange cannot see it as a valid number as well ... ? |
| 18:36.59 | E-Rage | jmls: Actually it can: when plugging directly into the PSTN, calls work fine. I think the receiving switch sees that the call is marked as international and routes it accordingly |
| 18:37.11 | robin_sz | Hmmm .. weird ... I plugged my HFC based ISDN card into the box, rebooted it, rant zttool .. and it can't see it!! ... |
| 18:37.17 | E-Rage | Receiving switch being the telco, that is. |
| 18:37.26 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 18:38.21 | jmls | Nah, don't think so .. I don't think there is such a thing as an "international" flag. could be wrong |
| 18:38.46 | jmls | What's more likely is that the zap config is stripping the prefix before it hits the * dialplan |
| 18:39.04 | robin_sz | nettie: ping |
| 18:39.30 | robin_sz | sigh .. so i have to do the bristuff thing before I can use a HFC card, right? |
| 18:39.49 | *** join/#asterisk krondorl (n=krondorl@207.245.14.10) |
| 18:39.52 | jmls | what's your zaptel.conf in /etc ? |
| 18:40.26 | jmls | and the zapata.conf in /etc/asterisk ? |
| 18:41.21 | E-Rage | jmls: zapata.conf: http://pastebin.ca/192152 |
| 18:43.12 | jmls | E-Rage: seems ok |
| 18:43.28 | jmls | what's the console output for a call ? and what's the dialplan ? |
| 18:45.35 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 18:46.30 | E-Rage | Diaplan is simple: exten -> _X.,1,Dial(to-pstn-pri......) |
| 18:47.02 | jmls | E-Rage: let us see the context and full dial command |
| 18:48.08 | robin_sz | tzafrir: you around? |
| 18:48.29 | E-Rage | jmls: http://pastebin.ca/192159 |
| 18:48.54 | E-Rage | where ${TRUNKPSTN} is "Zap/g1" |
| 18:50.26 | jmls | E-Rage: do me a favour, and try moving the group=1 directly above channel => 1-11 |
| 18:50.46 | jmls | and group=2 directly above channel => 25-35 |
| 18:51.04 | jmls | I've got a feeling ... |
| 18:51.18 | jmls | obviously restart asterisk and zap |
| 18:52.02 | E-Rage | jmls: should I move channel=> to directly below group=, or move group above channel? (ie, shouldn't group= be on top?) |
| 18:52.37 | jmls | move group down to just above channel |
| 18:53.07 | E-Rage | ack, someone on the phone :) can't restart just yet |
| 18:54.16 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:54.55 | tessier | When lack of bandwidth on a line is not the problem is there any reason to believe that going with compression would lower jitter and packet loss? |
| 18:55.05 | jmls | e-rage: see http://www.automated.it/guidetoasterisk.htm, just after "# vi /etc/asterisk/zapata.conf" |
| 18:55.21 | Holos | Is there anyway to push MWI down to a phone? I have a call center, where agents use a follow me script to forward their extension to a phone, I'd like to push wheter they have a VM waiting or not. |
| 18:55.28 | robin_sz | tzafrir: dood, does you debian/rapid stuff have the bristuff/zaphfc to get my HFC based isdn card going? |
| 18:55.31 | tessier | We have strange unexplained packet loss at one of our offices and a coworker is suggesting we turn on some compression. But we have 3Mb links and never use more than a meg or so. |
| 18:56.26 | robin_sz | tessier: you need vlan tagging or QOS |
| 18:56.32 | robin_sz | or seperate networks |
| 18:56.49 | tessier | robin_sz: We are using QoS but we only have control over one end of the link. |
| 18:56.55 | jmls | tessier: or fix a broken switch ... |
| 18:56.58 | tessier | And that end never has a problem. |
| 18:57.11 | Holos | tessier: Is it a Internet Link? |
| 18:57.13 | tessier | jmls: It is going across the Internet. If there is a broken switch it is beyond our control. |
| 18:57.14 | Holos | or a private link? |
| 18:57.15 | tessier | Holos: Yes |
| 18:57.17 | tessier | Internet |
| 18:57.18 | robin_sz | well, compression will not help |
| 18:57.25 | tessier | robin_sz: I didn't think so. |
| 18:57.29 | robin_sz | vlans are the way to go |
| 18:57.31 | tessier | Just wanted to get a sanity check |
| 18:57.47 | robin_sz | try a openvpn tunnel between the two ends |
| 18:57.48 | Holos | tessier: Then you're out of luck.. the ISP/Internet is dropping packets, compression won't help. |
| 18:57.48 | tessier | robin_sz: If it were all on our own infrastructure we could set up a vlan. But it goes across the net. |
| 18:57.54 | Holos | Is on the same ISP? |
| 18:57.57 | tessier | robin_sz: That was my next thought... |
| 18:57.59 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 18:58.01 | tessier | Holos: Not on the same ISP |
| 18:58.09 | tessier | robin_sz: If we make a tunnel we can QoS both ends. |
| 18:58.10 | japerry | heya, I have two questions, I wonder if you guys have seen |
| 18:58.12 | robin_sz | then, if dropped packets based on content, they cant spoil it for you |
| 18:58.19 | elduffy | back |
| 18:58.30 | *** join/#asterisk frawd (n=francois@21.Red-83-32-41.dynamicIP.rima-tde.net) |
| 18:58.33 | japerry | One: Grandstream phones GXP2000, if you put an outside caller on hold, it will never pick the line up again |
| 18:58.35 | robin_sz | fixing the packet loss is the real answer |
| 18:58.36 | Flauto | plantroniccdr |
| 18:58.54 | Holos | tessier: Then you have no control / recourse. OpenVPN and UDP based VPN will help, but like Robin_SZ said, you need to fix the packet loss.. |
| 18:58.59 | robin_sz | japerry: I have a patch for that |
| 18:59.02 | japerry | Two: Some phones (not all) seem to not be able to use the call menu system. They enter 1 into the IVR and it does nothing |
| 18:59.07 | japerry | robin_sz: awesome! |
| 18:59.23 | japerry | ohhh and for the second one, these are outside callers |
| 18:59.29 | *** part/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 18:59.30 | robin_sz | japerry: you need a tool to install it in the phoen though ... |
| 18:59.41 | E-Rage | jmls: doesn't appear to have changed anything, though I've now noticed that I'm getting chan_zap setup_zap errors that are "Ignoring <config>" where <config> are params from my zapata.conf |
| 18:59.42 | japerry | robin_sz like what tool? |
| 18:59.54 | robin_sz | japerry: well, roughly .. hammer shaped |
| 18:59.59 | japerry | ol |
| 19:00.03 | japerry | grrr |
| 19:00.03 | E-Rage | jmls: I don't think due to this change, becuase I changed back and still get the same eorrs |
| 19:00.15 | japerry | so let me guess, this is a common problem? |
| 19:00.20 | tessier | james_: Sounds like a DTMF problem. |
| 19:00.23 | robin_sz | japerry: I share your despair. I too have GXP2000s. the probelm is this: |
| 19:00.26 | robin_sz | they are crap |
| 19:00.33 | jmls | e-rage: did you restart asterisk, or zaptel as well ? |
| 19:00.35 | syzygyBSD | When I try to call my cell from an automated script it hangs up when it should transfer to voicemail, http://pastebin.ca/192170, however if I do it from a phone it works fine |
| 19:00.36 | elduffy | i'm experiencing a strange issue with suse and zaprtc. anybody familiar with that? |
| 19:00.42 | japerry | robin_sz: they used to work fairly well |
| 19:00.57 | robin_sz | japerry: this was before the software "upgrade" right? |
| 19:01.01 | japerry | and I know at one point 'hold' used to work... but not anymore |
| 19:01.06 | japerry | robin_sz: yup |
| 19:01.11 | tessier | elduffy: Based upon your incredibly detailed problem description I am sure we can all relaet. |
| 19:01.12 | tessier | relate |
| 19:01.13 | E-Rage | jmls: no, but I will once my calls complete |
| 19:01.17 | japerry | robin_sz: we had to upgrade to get the sidecar to work |
| 19:01.26 | robin_sz | japerry: I fell for that trick too. now they are not even very good doorstops. |
| 19:01.31 | tessier | Snom phones are da bomb btw |
| 19:01.33 | robin_sz | too light to hold a door open |
| 19:01.38 | jmls | you need to restart zaptel if you change zaptel.conf in /etc |
| 19:01.38 | tessier | I am not going to use anything else for quite a while. |
| 19:01.40 | Bpedersen | lol |
| 19:01.43 | tzafrir_home | BTW: "michael" the spammer of the voip-info wiki continues |
| 19:01.45 | japerry | robin_sz: lol ...... |
| 19:01.46 | jmls | service zaptel restart |
| 19:01.46 | tessier | Last year I spent way too much time fscking around with Cisco |
| 19:02.11 | [TK]D-Fender | sudhir492: Problem : you broke your ALERTINFO into 2 seperate tags. you need to do them in *1*. |
| 19:02.13 | tessier | Two years ago I was a total asterisk n00b...now I have deployed a number of systems. |
| 19:02.13 | japerry | well I suppose we could Ebay all the grandstream phones and try something else |
| 19:02.15 | elduffy | tessier: i can compile zaprtc successfully but am unable to "make load" it because port 112 is in use. lsmod doesn't show me any rtc module in use |
| 19:02.19 | robin_sz | japerry: seriously, I have three. none of them are even plugged in any more. the upgrade rendered them close to useless |
| 19:02.22 | japerry | but thats not probably going to fly with management |
| 19:02.30 | tessier | I should probably repay my karmic debt by answering questions in here. This channel helped me out a whole ton back in the day... |
| 19:02.38 | japerry | robin_sz: grr hmm ok |
| 19:02.46 | Bpedersen | robin_sz: what hw rev? |
| 19:02.47 | tessier | elduffy: port 112? Kernel modules don't use ports in the tcp or udp sense... |
| 19:02.48 | sudhir492 | D-Fender: thanks. |
| 19:02.50 | japerry | robin_sz: and you can't downgrade firmware if I remeber right |
| 19:03.00 | robin_sz | they appear to work on the surface, but too much is broken ... blanking screens, bad audio, random crashing ...and right, no way back :( |
| 19:03.12 | robin_sz | so ... |
| 19:03.12 | tzafrir_home | if you changes zaptel.conf you basically need to re-run ztcfg ... |
| 19:03.18 | tzafrir_home | Nothing more ... |
| 19:03.20 | robin_sz | back to this hammer I was talking about |
| 19:03.36 | jmls | tzafrir_home: cool |
| 19:03.49 | elduffy | tessier: i know... it's the IO port 0x70 which is used for communication w/ rtc |
| 19:03.55 | tzafrir_home | elduffy, why do you need zaprtc? do you use kernel 2.4? |
| 19:04.02 | tessier | elduffy: What is the exact error message you are receiving? |
| 19:04.02 | robin_sz | tzafrir_home: dood, do you have sarge debs for stuff to get an HFC based ISDN card going on * ? |
| 19:04.14 | *** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org) |
| 19:04.24 | elduffy | tzafrir_home: yes i have to... no uhci in place too |
| 19:04.30 | tzafrir_home | robin_sz, for zaphfc? |
| 19:04.43 | E-Rage | jmls: I've restarted: no change |
| 19:04.45 | robin_sz | tzafrir: is that what I need? .. then yeah |
| 19:04.46 | japerry | robin_sz: I wish there was more documentation about this before we made the plunge |
| 19:04.58 | tessier | I am so glad the dependency on a usb chipset for conference bridge timing etc was removed and replaced by kernel code. That was a PITA. |
| 19:04.59 | elduffy | tessier: syslog shows "rtc: I/O port 112 is not free." |
| 19:05.07 | tzafrir_home | robin_sz, that's what I have |
| 19:05.15 | robin_sz | japerry: yeah .. I made the plunge when the dire warning was on the bottom of the page, but it soon moved to the top |
| 19:05.26 | robin_sz | tzafrir_home: url? |
| 19:05.35 | japerry | robin_sz: I have hardware rev .4 |
| 19:05.39 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 19:05.42 | tzafrir_home | deb http://updates.xorcom.com/rapid sarge main |
| 19:05.53 | robin_sz | japerry: I just have a pile of them. dead. |
| 19:06.14 | stefmtl | hello, I have the trunk version, but chan_zap does not compile. What do I miss ? |
| 19:06.33 | tessier | elduffy: Interesting...never seen that before. It would suggest to me that you already have rtc capability. |
| 19:06.41 | tzafrir_home | stefmtl, we miss. The error you get, and other relevant information |
| 19:06.44 | tessier | elduffy: Are you sure you need that module loaded? |
| 19:06.55 | elduffy | tessier: to mee too, but lsmod doesn't show anything rtc related |
| 19:07.03 | tessier | elduffy: it may not be a module |
| 19:07.07 | tessier | elduffy: What kernel version? |
| 19:07.08 | Bpedersen | robin_sz: We dont really have that many problems with the gxp-2000 but thats rev 1.1 tou |
| 19:07.10 | elduffy | tessier: yes, i have to do trunking and meetme |
| 19:07.16 | Bpedersen | only headsets are a pain |
| 19:07.17 | elduffy | tessier: 2.4 |
| 19:07.23 | Bpedersen | mic gain is way to low |
| 19:07.30 | japerry | Bpedersen: we have some rev 1.1s too |
| 19:07.37 | japerry | with the green lights |
| 19:07.39 | elduffy | tessier: how do i determine if it is something else? |
| 19:07.50 | Bpedersen | japerry: yeah |
| 19:07.50 | robin_sz | tzafrir_home: so I dont need to patch the kernel or anything crazy like that? |
| 19:07.51 | stefmtl | elduffy : I have no compile error, asterisk is working fine, except I don't have a chan_zap |
| 19:07.55 | Holos | Is there anything that causes the internal db1 database "database show" to loose it's entries? |
| 19:08.05 | tzafrir_home | robin_sz, no |
| 19:08.07 | japerry | the funny thing is, that when you put someone on hold internally, it works, externally it doesn't |
| 19:08.23 | japerry | I'm wondering, is the hold issue an asterisk issue perhaps? |
| 19:08.30 | Bpedersen | japerry: i dont have that problem at all |
| 19:09.18 | japerry | hmm you can call or be called externally, put someone on hold, and it doesn't sieze.. okay. |
| 19:09.26 | japerry | ooh what firmware you running? |
| 19:09.44 | robin_sz | tzafrir: ok, great. thanks |
| 19:09.57 | Bpedersen | was tricked into using the 1.1.1.14 by grandstream to try and fix our mic gain problems |
| 19:10.03 | robin_sz | tzafrir_home: wait, I dont see a zaphfc package ... whats it in? |
| 19:10.16 | japerry | Bpedersen: and did that f' more things up? |
| 19:10.28 | robin_sz | Bpedersen: I think its a cunning plan to destroy old phones ... |
| 19:10.37 | Bpedersen | japerry: it didnt help on the gain thats for sure hehe |
| 19:10.44 | tzafrir_home | zaphfc is one of the modules in zaptel-modules-`uname -r` |
| 19:10.50 | Bpedersen | robin_sz: nod |
| 19:10.51 | japerry | Bpedersen.. but what stopped working? ;-) |
| 19:11.07 | elduffy | tessier: ? |
| 19:11.35 | Bpedersen | japerry: sometimes when you answed 2 or 3 call the mic doesnt "unmute" |
| 19:11.42 | robin_sz | tzafrir_home: ahh .. so I might get problems combining that with my built from latest sources zaptel-1.2.9.1? |
| 19:12.21 | tessier | elduffy: Have you tried doing meetme? Does it not work? |
| 19:12.24 | Bpedersen | japerry: and the headset mic gain is worse then before |
| 19:12.35 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:12.43 | tzafrir_home | It must match the version of zaptel. It uses exports from zaptel |
| 19:12.44 | elduffy | tessier: it can't as no zap is available |
| 19:12.52 | Bpedersen | robin_sz: what kind of phones did you change to? |
| 19:12.57 | tzafrir_home | It will probably fail to load otherwise |
| 19:12.59 | robin_sz | Bpedersen: I found that replacing that lump on the end of ethernet cable helped a lot |
| 19:13.02 | robin_sz | oh, Snom |
| 19:13.07 | robin_sz | 320s and 360s |
| 19:13.08 | *** part/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
| 19:13.26 | japerry | damn |
| 19:13.28 | japerry | they're so much more |
| 19:13.34 | japerry | but alas they work |
| 19:13.35 | japerry | ? |
| 19:13.43 | robin_sz | the Snom300 is good value |
| 19:13.45 | tzafrir_home | robin_sz, alternatively, take the source from zaptel-source and build it manually |
| 19:13.46 | Bpedersen | have anyone tried aastra phones? |
| 19:14.20 | robin_sz | tzafrir_home: how does that differ from my zaptel-1.2.9.1 source, patched in some way? |
| 19:14.32 | tzafrir_home | If you don't like packages. But all the zaptel modules *must* come from the same build |
| 19:15.41 | robin_sz | oh I DO like packages ... but for some reason I ended up compiling stuff from source .. I forget why |
| 19:15.53 | tzafrir_home | robin_sz, IIRC the kernel adds versioning to exported symbols, and thus changes to the source and build environment cause changes to the actual symbol |
| 19:16.13 | robin_sz | probably because the Sarge packages were ancient, so I built it myself |
| 19:16.48 | E-Rage | jmls: I set pri debugging on, and the number does come through marked as an international number (TON) |
| 19:17.10 | E-Rage | jmls: But when I dial the number out to the PSTN, the TON is "unknown" (perhaps expected) |
| 19:17.16 | E-Rage | Per pridialplan |
| 19:17.42 | japerry | so |
| 19:17.50 | japerry | Get this guys---it goes on hold if an outside caller calls in |
| 19:18.02 | japerry | but if the grandstream makes the outgoing call, and puts on hold, it doesn't work |
| 19:18.09 | japerry | I'm starting to guess, its not the phone |
| 19:18.14 | *** join/#asterisk CyberPony (n=CyberPon@cpe-071-075-174-216.carolina.res.rr.com) |
| 19:19.33 | CyberPony | I'm experiencing what appears to be a clocking issue with 2 T1 PRI's on a single FX110P card |
| 19:19.52 | tzafrir_home | We're currently "stuck" with 1.2.8 in our development branch because it has a few customizations and it will take a while to move them properly in the local SVN |
| 19:19.53 | CyberPony | does anyone have any experience with this card? |
| 19:20.07 | japerry | OMG |
| 19:20.13 | japerry | okay, so this 'fixed' the issue |
| 19:20.22 | japerry | I had to use the transfer function |
| 19:20.28 | japerry | it put the caller on hold with music and all |
| 19:20.36 | japerry | then now EVERY phone works with hold |
| 19:20.43 | japerry | I think its an asterisk issue, but not sure where |
| 19:24.56 | syzygyBSD | I put the laughter in manslaughter |
| 19:25.17 | *** join/#asterisk aadilismail (n=aadilism@202.125.143.70) |
| 19:25.42 | aadilismail | how to change port 5060 ? |
| 19:25.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:26.05 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 19:26.08 | syzygyBSD | aadilismail: in sip.conf |
| 19:27.01 | *** part/#asterisk toxap (n=toxap@213.227.193.75) |
| 19:27.44 | aadilismail | bindport? |
| 19:28.31 | syzygyBSD | think so |
| 19:28.45 | syzygyBSD | no, just port |
| 19:28.47 | Jason99 | I'm making outgoing .call files and for the channel I have Local/XXXXXXXXXX@my-context, the call is dialed according to the my-context. When the call is answered, it does a Playback(file) but I get the following in VERBOSE, WARNING[490]: chan_sip.c:2561 sip_write: Asked to transmit frame type 2, while native formats is 4 (read/write = 64/2) |
| 19:28.54 | syzygyBSD | port = 5060 ; Port to bind to |
| 19:29.18 | aadilismail | ok thanx |
| 19:30.21 | robin_sz | tzafrir_home: so making sure zaptel etc were not loaded in the kernel right now, installing that .deb and then starting * should do it? |
| 19:30.26 | syzygyBSD | Jason99: do you see those errors any other time when making a call to that extension? |
| 19:30.28 | hwt | hm, every once in a while, when i hang up after a short call, asterisk calls me back. |
| 19:30.39 | robin_sz | hwt: it likes you. |
| 19:30.41 | hwt | even if the call is a pstn call through another sip gw. |
| 19:30.48 | syzygyBSD | hwt how short of a call |
| 19:30.49 | tzafrir_home | /etc/init.d/zaptel unload |
| 19:30.49 | hwt | it says "asterisk" in the display. :) |
| 19:30.57 | hwt | syzygyBSD: perhaps 0-3 seconds. |
| 19:30.58 | tzafrir_home | or: genzaptelconf -u |
| 19:31.05 | hwt | asterisk as the callid, even. |
| 19:31.11 | hwt | is this a bug or a feature? |
| 19:31.12 | syzygyBSD | hwt I have seen that before with some sip gateways |
| 19:31.28 | hwt | ok, so this is a bug? |
| 19:31.33 | tzafrir_home | hwt, you didn't really hang up: you've just flashed |
| 19:31.41 | tzafrir_home | right? |
| 19:31.57 | syzygyBSD | ya... think that is the way the sip gateway thinks |
| 19:32.05 | hwt | tzafrir_home: yeah, that might be! |
| 19:32.06 | robin_sz | tzafrir_home: I have stop|start|restart|force-reload :( |
| 19:32.14 | hwt | tzafrir_home: so what is this? |
| 19:32.38 | tzafrir_home | press on the phone's button for a few seconds |
| 19:32.53 | robin_sz | I think my Sipura SPA2102 may be worse than my GXP2000 |
| 19:33.01 | robin_sz | hard to beleive .. but ... |
| 19:33.16 | tzafrir_home | robin_sz, well, those are just equivalents of rmmods . But rmmods will do just fine |
| 19:33.21 | hwt | tzafrir_home: it's hard to reproduce on my dect phone. i can test it at the office with a regular phone tomorrow. |
| 19:33.34 | robin_sz | tzafrir_home: right ... |
| 19:33.35 | hwt | tzafrir_home: what do you mean press the button for a few seconds. |
| 19:33.37 | hwt | tzafrir_home: ? |
| 19:33.58 | tzafrir_home | hwt, to make sure you don't actually flash |
| 19:34.07 | hwt | tzafrir_home: i probably flash. |
| 19:34.29 | hwt | tzafrir_home: since it's hard to reproduce on the dect. (i cant pick up and hang up fast enough) |
| 19:34.44 | hwt | so is there a default event to call back when i flash+ |
| 19:35.28 | syzygyBSD | well, a flash isn't a hangup, so it still tries to send the call to you, I was able to reproduce that with analog lines a while back |
| 19:35.46 | Jason99 | syzygyBSD: If I use SIP/, everything works fine.. but when I use Local/ it doesn't |
| 19:35.59 | hwt | syzygyBSD: ok, the asterisk version running on that box is getting old, so maybe i should upgrade it |
| 19:36.29 | Jason99 | syzygyBSD: The reason for the Local/ is so that my-context has some routing built in and decides which SIP gateway to send it to.. Maybe there is another way? |
| 19:38.14 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 19:38.16 | pifiu-laptop | anyone have the default music on hold of cisco unified messaging? aka Cisco call manager? |
| 19:38.32 | syzygyBSD | well, there is always anther way, but my guess as to why that is happening is that the local channel is using a different format than the SIP channel |
| 19:38.53 | syzygyBSD | Jason99: you are creating this with a call file right? |
| 19:39.07 | Jason99 | syzygyBSD: Yes |
| 19:39.22 | syzygyBSD | why do you need 3 contexts to do this? |
| 19:40.14 | syzygyBSD | can you pastbin your call file and the 3 contexts you are using? |
| 19:40.24 | syzygyBSD | I'll cut it down a little |
| 19:40.24 | Jason99 | I have one context to decide where to route the call and the other context is called when the call is answered |
| 19:40.41 | syzygyBSD | there is another one in there too... |
| 19:42.10 | Jason99 | syzygyBSD: no, just the two.. the first query mysql to find the route for the specific prefix and dials the call.. that part is successful.. as soon as I answer the ringing phone, i get that error on the console when it tries to play |
| 19:42.11 | syzygyBSD | Just pastebin your call file please |
| 19:42.17 | Jason99 | sure |
| 19:44.20 | Jason99 | syzygyBSD: http://pastebin.ca/192220 |
| 19:45.06 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:46.51 | robin_sz | tzafrir_home: ok, done :) ... now next question |
| 19:47.15 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 19:47.30 | robin_sz | tzafrir_home: what tests can I do, in advance of my telco changing the line to ISDN, to make sure it is probably going to work? |
| 19:47.37 | elriah | Hey guys - anyone ever seen a situation where g729 decoders wouldn't free themselves back up after being used once? (1.2.12.1) |
| 19:48.17 | [TK]D-Fender | sudhir492: So... working now? |
| 19:48.21 | tzafrir_home | robin_sz, sorry. This is something you'll have to ask people from BRI-land |
| 19:48.25 | ernie_ | is it possible that iax2 has a worse sound quality than sip? |
| 19:48.26 | tzafrir_home | (europe) |
| 19:48.37 | ernie_ | same codecs/same provider |
| 19:48.44 | *** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) |
| 19:48.44 | hwt | ernie_: the, no. |
| 19:48.47 | diablopico | hello |
| 19:48.48 | hwt | ernie_: then, no. |
| 19:49.02 | robin_sz | tzafrir_home: I tried setting loopback in zttool and it just hunh there ... |
| 19:49.08 | tzafrir_home | robin_sz, where I live, the telco it still POTS |
| 19:49.09 | hwt | ernie_: unless you have shaping or something on the IAX port. |
| 19:49.25 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:49.32 | ernie_ | i have shaping on it, i put it on high priority? |
| 19:49.35 | diablopico | does anyone know what module to load for the te212p card after modprobe zaptel ? |
| 19:49.39 | robin_sz | tzafrir_home: that must be somewhere deep in darkest peru? |
| 19:49.56 | ernie_ | while my sip traffic was still in normal priority |
| 19:50.19 | tzafrir_home | diablopico, the module of the card itself. And it will actually load zaptel on its own |
| 19:50.39 | aadilismail | how to change port=5060 in asterisk .... ? |
| 19:50.50 | robin_sz | aadilismail: in sip .conf |
| 19:50.53 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.155.94) |
| 19:51.00 | tzafrir_home | diablopico, why not try xpp/utils/genzaptelconf -d and you'll see immedietly .... |
| 19:51.24 | robin_sz | aadilismail: with a text editor is the easiest way |
| 19:51.44 | diablopico | thanks |
| 19:52.39 | Jason99 | syzygyBSD: I figured it out |
| 19:52.48 | syzygyBSD | what was it |
| 19:53.01 | syzygyBSD | sorry, boss pulled me away talking about a client |
| 19:53.03 | aadilismail | thanx |
| 19:53.12 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 19:53.32 | Jason99 | syzygyBSD: If you want the Local/ channel to act exactly as a normal channel you need to do (ex: Local/XXXXXXXXXX@my-context/n) |
| 19:53.41 | Jason99 | you need /n at the end |
| 19:53.46 | syzygyBSD | ahh |
| 19:55.50 | sudhir492 | D-Fender: Unfortunately not yet |
| 19:56.23 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 19:58.18 | sudhir492 | D-Fender: Do you have a working example? |
| 19:59.18 | robin_sz | so .. having modprobed my HFC zaptel thing and zttool can see the card .. I need to change /etc/zaptel.conf in some random way, right? |
| 20:00.43 | tzafrir_home | robin_sz, without actually connecting to the telco, there is no sync in the line and the line should have some sort of alarm, right? |
| 20:01.30 | robin_sz | mmm ... no. it says OK, an there is no line yet ... |
| 20:02.08 | robin_sz | I have a X100P in the box as well ... and fxoks=1 in my zaptel.conf |
| 20:03.02 | robin_sz | so just fxoks=1-3 will do? |
| 20:03.10 | robin_sz | kewlstart for cannels 1 to 3? |
| 20:03.40 | *** join/#asterisk xnon (n=xnon@200.8.87.1) |
| 20:08.11 | pifiu-laptop | damn cisco people are d*cks! |
| 20:08.24 | aydiosmio | yes we are! |
| 20:08.40 | pifiu-laptop | yes! |
| 20:08.41 | pifiu-laptop | lol |
| 20:08.47 | pifiu-laptop | well the people in that channel at least |
| 20:09.09 | krondorl | I was wondering if anyone knew how to change the recording time default when someone is leaving a message and if it is possible to send them back to the prompt system (or warn them) when their time is running out?? |
| 20:09.30 | krondorl | has run out |
| 20:11.53 | *** join/#asterisk _julian (n=julian@dslb-082-083-131-183.pools.arcor-ip.net) |
| 20:11.57 | _julian | hi all |
| 20:12.53 | _julian | is it possible to forward an incoming ISDN (capi) call to a SIP phone, without answering the call until the SIP phone answers? - so that it keeps rining on a normal isdn-phone that's also connected to the line? |
| 20:16.44 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.184.62) |
| 20:17.32 | *** join/#asterisk MattH (n=MattH@cloud2.chilitech.net) |
| 20:17.37 | *** part/#asterisk _julian (n=julian@dslb-082-083-131-183.pools.arcor-ip.net) |
| 20:18.05 | MattH | Greetings.. I am attempting to trouble shoot a digium card... should or should not there be interrupt hit numbers increasing on /proc/interrupts if the card is working correctly? |
| 20:18.24 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-6d4cf5e6d9b775d2) |
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| 20:21.24 | syzygyBSD | _julian yes |
| 20:22.47 | syzygyBSD | MattH: no, the interuppts should stay the same |
| 20:23.24 | syzygyBSD | I belive for digium cards it is 1000 |
| 20:23.40 | bungalow | hi all - I've got a strange problem where after about 18-20 minutes, sometimes more into a call, DTMF is no longer reported. Using app_conference to show DTMF events in the manager, and asterisk 1.2.10. I've checked and verified through port cap that the DTMF is coming in correctly -- even when not reported. Has anyone heard anything about such a problem? |
| 20:23.52 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
| 20:24.33 | bungalow | after hanging up and reconnecting, DTMF works again |
| 20:24.34 | MattH | ok so interrupts should not be increasing? |
| 20:25.01 | syzygyBSD | not for the digium card, no |
| 20:25.21 | MattH | hrmm ok.. the wiki says it should be increasing... |
| 20:25.29 | syzygyBSD | what page? |
| 20:25.38 | MattH | http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshootin |
| 20:25.56 | MattH | Make sure your Digium |
| 20:25.57 | MattH | hardware is on its own IRQ by itself and that it is taking interrupts. |
| 20:26.16 | aadilismail | how to convert port=5060 to port=8002......? |
| 20:29.15 | syzygyBSD | matth the next line is this should not be 0 |
| 20:29.34 | syzygyBSD | dang people that leave.... |
| 20:29.59 | syzygyBSD | aadilismail: in sip.conf add the line 'port = 8002' |
| 20:30.04 | syzygyBSD | without the ' |
| 20:30.10 | hmmhesays | yes, my prepaid app is working well |
| 20:30.33 | *** join/#asterisk ToxaP (i=ToxaP@194.187.128.88) |
| 20:30.50 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-197-7.red.bezeqint.net) |
| 20:32.25 | bungalow | does anyone know if there is a way to output DTMF events to the console? |
| 20:32.35 | *** join/#asterisk stephane_ (n=stephane@2001:6f8:361:10:205:2ff:fede:c350) |
| 20:32.35 | bungalow | for debug purposes? |
| 20:34.00 | *** join/#asterisk Thus0 (n=Thus0@169.111.102-84.rev.gaoland.net) |
| 20:34.05 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 20:35.19 | ToxaP | bungalow, in debug mode all information is a way to output |
| 20:35.48 | DoDaT69 | where can I set the extension for the analog channel? zapata.conf/ |
| 20:35.53 | *** join/#asterisk toxap (i=toxap@194.187.128.88) |
| 20:36.05 | bungalow | does anyone understand what ToxaP means? |
| 20:37.04 | aydiosmio | I imagine is some sort of codeword for "confused" |
| 20:38.12 | bungalow | is he trying to say that there is a way to get DTMF events reported to the CLI in debug mode? |
| 20:39.23 | mog | yes bungalow |
| 20:39.26 | mog | in logger.conf |
| 20:39.35 | mog | console |
| 20:39.38 | mog | dtmf |
| 20:39.55 | bungalow | ahhh... great, thanks a lot |
| 20:40.53 | Yogik | Guys , is there any reason asterisk would not detect that SIP trunk hung up the line? |
| 20:41.11 | *** join/#asterisk Renacor (n=kvirc@66.238.64.20) |
| 20:41.22 | Renacor | is there a way to specify ZAP/extension# ? |
| 20:43.14 | bungalow | mog: I've got a strange problem, where after several minutes (often 20 or more) asterisk stops reporting DTMF on a given channel -- if I reload, it starts working again. any idea how I can debug this? |
| 20:43.17 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:43.32 | mog | id you dtmf debug |
| 20:43.57 | *** join/#asterisk snickl (n=snickl@dialin126151.server4you-dsl.de) |
| 20:46.04 | mog | er use dtmf debug |
| 20:46.52 | *** join/#asterisk mmurdock (n=vircuser@mail.kimballequipment.com) |
| 20:49.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:50.42 | file | [TK]D-Fender: home! |
| 20:51.05 | [TK]D-Fender | 1nd33d |
| 20:52.18 | hmmhesays | cool |
| 20:52.28 | hmmhesays | how's that for you file? |
| 20:52.49 | file | what is what for me? |
| 20:52.50 | hmmhesays | and I need a good recommendation for a multifunction printer |
| 20:52.58 | hmmhesays | print fax scan |
| 20:53.39 | sevard | are you printing color? |
| 20:53.41 | [TK]D-Fender | hmmhesays : what scale? |
| 20:53.58 | hmmhesays | small, for my office here |
| 20:54.02 | hmmhesays | sevard: don't have to |
| 20:54.03 | sevard | office max had a sale on hp officejet 5610s a couple weeks ago |
| 20:54.09 | sevard | it was cheap and works pretty slick |
| 20:54.20 | hmmhesays | you have one? |
| 20:54.26 | sevard | yeah |
| 20:55.02 | sevard | i've only used it a total of 6 times though, so it's not a very good review |
| 20:55.12 | sevard | it was like 70 bucks with all the rebates and shit |
| 20:55.15 | [TK]D-Fender | hmmhesays : Might suggest away from multifunction if its for small usage |
| 20:55.17 | robin_sz | sigh ... OK, so what should I put in zaptel.conf to make a X100P and a HFC ISDN card play nicely ??? |
| 20:55.34 | [TK]D-Fender | hmmhesays: the cost effectiveness of the cartidges can really blow and cost you in the end |
| 20:55.44 | sevard | [TK]D-Fender: I too hate multifunctions and that's a valid point |
| 20:55.48 | hmmhesays | [TK]D-Fender: i'm going to check the price of this 5610 cartridges |
| 20:55.55 | hmmhesays | and if I can use el cheapo refill ink |
| 20:55.56 | file | I bought my laser printer for $140 3 years ago and I'm still on the same toner as then... works great |
| 20:56.08 | sevard | i can't remember how much the carts are |
| 20:56.13 | robin_sz | old HP laserjets are great for B&W ... cheap carts |
| 20:56.21 | sevard | file: nice |
| 20:56.23 | robin_sz | and good printers for like $50 on ebay |
| 20:56.35 | sevard | file: my old laserjet has been telling me for about 6 months now that it's out of ink |
| 20:56.36 | robin_sz | for colour, it has to be Xerox Phaser |
| 20:56.41 | sevard | file: still prints like a dream though |
| 20:56.42 | [TK]D-Fender | hmmhesays : Look at the Officejet K550DTN. Its a great printer and its cheaper per page than most lasers and doe Duuplex, 2 trays (for fixed letter & legal), and is networked. All for under $300. |
| 20:56.45 | file | sevard: haha |
| 20:57.03 | hmmhesays | i need a fax too though |
| 20:57.04 | sevard | so don't trust thoes meters |
| 20:57.08 | robin_sz | hmmhesays: send or rx? |
| 20:57.11 | sevard | i print on a daily basis too |
| 20:57.18 | [TK]D-Fender | sevard : if you HAVE the volume, the HP 4345 MFP is an amazing printer/scanner/fax. |
| 20:57.38 | sevard | [TK]D-Fender: you can always pick up old okidatas from highschools and print on reams :D |
| 20:57.43 | sevard | reems? |
| 20:57.47 | [TK]D-Fender | sevard : I run 3 at my job. Photocopy as well obviousy.... very good at it too |
| 20:58.00 | [TK]D-Fender | sevard : Oh yeah, we run reams through it.... |
| 20:58.23 | [TK]D-Fender | sevard : Customer service fax, prints all our invoices, photocopys too..... loaded |
| 20:58.24 | hmmhesays | robin_sz: both |
| 20:58.46 | robin_sz | hmmhesays: fair enuff .. you need a fax then |
| 20:59.39 | [TK]D-Fender | hmmhesays : Well just calc out the cost/page for each thing you'll do, add up the cost of the single/multiple machine combo's that look viable and you'll come up with an answer... |
| 21:00.11 | hmmhesays | yeah |
| 21:00.17 | hmmhesays | the hp 5610 looks ok |
| 21:00.19 | hmmhesays | and its on sale |
| 21:00.24 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
| 21:00.24 | *** mode/#asterisk [+o kram] by ChanServ |
| 21:02.03 | toerkeium | hello guys, when I try to "make linux26" for the zaptel module, I get an error telling that my kernel configuration didn't have modules enabled. So, I need to compile a custom kernel? I installed the last CentOS release, but I never make a custom kernel.. I wonder if I can enable modules from the installation process |
| 21:02.06 | toerkeium | any help? |
| 21:02.06 | [TK]D-Fender | hmmhesays : I jsut read the specs.... I am NOT liking those ink cartridges.... |
| 21:02.31 | *** join/#asterisk Joel1978_ (n=Joel1978@12-226-85-195.client.mchsi.com) |
| 21:02.31 | robin_sz | this is the thing I dont get .. I have two cards ... an X100P and a HFC isdn .. I set span=2,1,3,ami,ccs |
| 21:02.43 | *** part/#asterisk Joel1978_ (n=Joel1978@12-226-85-195.client.mchsi.com) |
| 21:02.43 | robin_sz | which should set span2 right? |
| 21:02.56 | *** join/#asterisk Joel1978_ (n=Joel1978@12-226-85-195.client.mchsi.com) |
| 21:03.05 | hmmhesays | [TK]D-Fender why not? |
| 21:03.24 | robin_sz | and then fxsks=1 |
| 21:03.36 | robin_sz | bchan=2-3 |
| 21:03.40 | robin_sz | dchan=4 |
| 21:03.44 | robin_sz | <PROTECTED> |
| 21:03.48 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 21:04.51 | [TK]D-Fender | hmmhesays : The large ones are <20ml. |
| 21:05.01 | [TK]D-Fender | hmmhesays : cost it out over the K550DTN. |
| 21:05.20 | robin_sz | but then i get: ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 21:05.20 | robin_sz | FATAL: Error running install command for wcfxo |
| 21:05.24 | robin_sz | sigh |
| 21:05.58 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:06.13 | hmmhesays | estimated 450 page yield |
| 21:06.16 | hmmhesays | that isn't bad |
| 21:06.58 | tzafrir_home | robin_sz, dump that install command |
| 21:07.17 | tzafrir_home | you have a proper zaptel init.d script that runs ztcfg, right |
| 21:07.19 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 21:07.19 | hmmhesays | [TK]D-Fender looks like that k550dtn has the same volume print cartridge |
| 21:07.32 | diclophis-work | so whats with asterisk and jabber? |
| 21:08.01 | robin_sz | tzafrir_home: yes .. I do |
| 21:08.28 | robin_sz | tzafrir_home: this is waht happens with /etc/init.d/zaptel start |
| 21:08.33 | tzafrir_home | robin_sz, or better yet, get rid of hte need to run ztcfg: http://bugs.digium.com/view.php?id=7613 |
| 21:09.36 | tzafrir_home | wcfxo is channel 2? shouldn't it be channel 1? |
| 21:09.42 | robin_sz | how would I know? |
| 21:09.51 | tzafrir_home | cat /proc/zaptel/* |
| 21:09.51 | [TK]D-Fender | hmmhesays : and the 5610 is a tri-color cartidge as opposed to seperate. |
| 21:10.07 | tzafrir_home | that's basically how genzaptelconf knows |
| 21:10.43 | robin_sz | genzaptelconf? |
| 21:11.05 | hmmhesays | [TK]D-Fender: thats not what i'm reading |
| 21:11.10 | tzafrir_home | robin_sz, maybe when you first added it, asterisk was running and had a pseudo-channel that was using channel no. 1 |
| 21:11.11 | hmmhesays | it uses the hp56 ink cartridge |
| 21:11.15 | hmmhesays | black |
| 21:11.38 | robin_sz | it had a wcfxo on 1, I added a HFC which should be on t ... |
| 21:11.39 | robin_sz | 2 |
| 21:12.07 | robin_sz | and indeed it is |
| 21:13.01 | robin_sz | I suspect I need a span=1,<something> as well then? |
| 21:14.10 | [TK]D-Fender | hmmhesays : Low capacity page yield (colour) |
| 21:14.10 | [TK]D-Fender | C9352AE HP 22 Tri-colour Inkjet Print Cartridge (5 ml) 138 pages (+/- 10%)* |
| 21:14.23 | hmmhesays | [TK]D-Fender: i'm going to be doing mostly black and white |
| 21:14.27 | hmmhesays | I can use the hp 56 |
| 21:14.35 | [TK]D-Fender | Page yield (photo) |
| 21:14.35 | [TK]D-Fender | C6658AE HP 58 Photo Inkjet Print Cartridge (17 ml) 125 photos (10 x 15 cm)* |
| 21:14.41 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 21:14.52 | [TK]D-Fender | hmmhesays : Was just quoting the colour cost |
| 21:14.57 | hmmhesays | ahh gotcha |
| 21:15.04 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 21:15.09 | pablus | hmm |
| 21:15.14 | pablus | affternoon |
| 21:15.29 | robin_sz | hmm, genzaptelconf isnt exactly .. umm, verbose is it? |
| 21:15.41 | robin_sz | callerid=asreceived |
| 21:15.43 | pablus | which sip client may you recommende to me for use asterisk? |
| 21:15.45 | robin_sz | and thats yer lot |
| 21:15.49 | TexasJay | Greetings. Is there a question-asking protocol I should follow or can I just blurt it out? :) |
| 21:16.03 | pablus | i use sjphone but i would like to test other but free |
| 21:16.06 | robin_sz | TexasJay: too late, you already asked to ask |
| 21:16.12 | TexasJay | ;) |
| 21:16.29 | robin_sz | fire away ... |
| 21:16.51 | TexasJay | Is it possible to dial an extension by its number? |
| 21:16.55 | robin_sz | yes |
| 21:17.08 | robin_sz | thats what the dialplan is for |
| 21:17.20 | TexasJay | Um, lemme see if I can reword it. :) |
| 21:17.31 | xheliox | Anyone know if the wanpipe stuff form Sangoma is working with Zaptel 1.4? |
| 21:17.51 | robin_sz | please do |
| 21:18.09 | robin_sz | exten => 5000,1,Macro(stdexten,5000,SIP/jaysphone) |
| 21:18.15 | [TK]D-Fender | hmmhesays : http://h10010.www1.hp.com/wwpc/za/en/ho/WF06c/A1-1782629-1782705-1782705-1782707-12229950-54738409.html |
| 21:18.50 | [TK]D-Fender | 820 pages est / cartridge (black) |
| 21:19.22 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 21:19.38 | [TK]D-Fender | hmmhesays : HP 56 (yours) Page yield (black and white, A4) |
| 21:19.38 | [TK]D-Fender | Approximately 450 pages at 5% coverage |
| 21:19.41 | tzafrir_home | xheliox, probably yes: zaptel 1.4 has changed very little from zaptel 1.2 (apart from the build system) |
| 21:19.52 | [TK]D-Fender | hmmhesays : HP 56 (yours) : |
| 21:19.58 | tzafrir_home | If you can get it to build, it will practically work the same |
| 21:20.01 | [TK]D-Fender | Page yield (black and white, A4) |
| 21:20.01 | [TK]D-Fender | Approximately 450 pages at 5% coverage |
| 21:20.08 | [TK]D-Fender | 1/2 .... ICK |
| 21:20.20 | cekc | is there a way to add delay before dialing the phone number on outgoing calls through my Digium T1 card? When I dial numbers such as 304-1123 it missing the 30 and calls 411 |
| 21:20.36 | robin_sz | thats nothing to do with delay |
| 21:20.50 | cekc | o |
| 21:20.59 | E-Rage | Anyone familiar with how to get PRI TON/NPI "dereferencing" working with pridialplan=dynamic? |
| 21:21.07 | robin_sz | thats some part of the dialplan or zapate.conf snipping off the first two digits |
| 21:22.40 | cekc | My log shows: Oct 5 13:40:25 DEBUG[27109] chan_zap.c: Dialing '3041123' |
| 21:23.28 | tzafrir_home | cekc, use , |
| 21:23.34 | tzafrir_home | sorry: w |
| 21:24.00 | diablopico | hello ,,, i have loaded drivers for te212p (wct4xxp) and it works , but only for 24 channels each span, doesnt this work on E! configurations ???? |
| 21:24.21 | [TK]D-Fender | diablopico : sure it does. Show us your configs |
| 21:24.36 | tzafrir_home | diablopico, isn't there a switch on the card for that? |
| 21:24.38 | [TK]D-Fender | diablopico : www.pastebin.ca |
| 21:24.40 | cekc | how do I use w ? |
| 21:24.54 | diablopico | let me look at that ,,, |
| 21:25.06 | tzafrir_home | cekc, as part of the number. IIRC it waits half a second |
| 21:25.43 | robin_sz | coo, it could be a delay then. my mistake |
| 21:26.15 | diablopico | thanks ,,, there be jumpers there !! |
| 21:36.48 | TexasJay | ok, sorry about the delay robin_sz. Got called away from my desk :D |
| 21:36.49 | diclophis-work | so whats this all about? "PRI got event: HDLC Abort (6) on Primary D-channel of span 1" |
| 21:37.34 | TexasJay | So, say I have a user Jay at the same extension (100) at the same phone. This never changes. Jay is always at 100 at that phone... |
| 21:37.36 | *** join/#asterisk _Syntax_ (n=Miranda@gw.sapientem.com) |
| 21:37.56 | TexasJay | I can call either 100 or 'jay' and it will ring just fine. |
| 21:38.15 | *** part/#asterisk _Syntax_ (n=Miranda@gw.sapientem.com) |
| 21:38.22 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 21:38.55 | TexasJay | Say I have the same setup with Robin and extension 101. Same phone, same Robin, same extension. |
| 21:39.20 | robin_sz | tzafrir: I think the line in modprobe is running ztcfg as each module is loaded, and so it runs ztcfg when the wcfxo module is loaded, but before the zaphfc module is loaded ... |
| 21:39.38 | TexasJay | I want to have users be able to dial a '6' and the extension number and have the phone enter into paging mode. |
| 21:39.57 | TexasJay | I already can get it to auto-answer so that's not the problem. |
| 21:40.25 | TexasJay | What I want is to have a pattern-matching extension that'll dial that extension. |
| 21:40.32 | TexasJay | Am I making any sense? |
| 21:40.45 | TexasJay | Dial 6 + ext; dial that ext in paging mode. |
| 21:41.17 | TexasJay | The phone auto answers. Like an intercom. |
| 21:41.39 | [TK]D-Fender | TexasJay : sURE YOU CAN GO THAT. |
| 21:41.44 | hmmhesays | so I wasn't around, does asterisk support t.38 passthru? |
| 21:42.34 | TexasJay | In my sip.conf file, I have users based on their name ([jay], [robin], [fender], et. al) and not by their extensions (101,102,103...) |
| 21:42.40 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 21:42.46 | raidenz | hi guys |
| 21:42.55 | [TK]D-Fender | TexasJay : ok, completely no impact on your auto-answer needs |
| 21:42.56 | TexasJay | So when someone dials 101, it's actually dialing SIP/jay |
| 21:43.59 | robin_sz | tzafrir_home: OK, the problem was that /etc/modprobe/zaptel had a && /usr/sbin/ztcfg on every line, removing it off the wcfxo line means ztcfg only gets run when the second (final) module is loaded .. problem solved |
| 21:44.09 | raidenz | It seems the REALTIME app/function in the dialplan has changed from 1.2 -> 1.4. I used to have Realtime(sippeers|name|5000|Variable) but thats not valid in 1.4. Is the realtime function in 1.4 now *only* return one value or the array like in 1.2? I tried exten => 1,1,Set(MyVariable=${REALTIME(sippeers|name|1000)}) and nothing is returned. ( even tried viewing all variables with dumpchan). Any ideas? |
| 21:46.32 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 21:48.05 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 21:50.13 | Beighto | Is Polycom still the best choice for IP phones? |
| 21:50.48 | anthonyl | Beighto, i think so |
| 21:51.19 | Beighto | cool |
| 21:52.43 | RoyK | anyone heard that the big asterisk users are considering openpbx? |
| 21:52.55 | [TK]D-Fender | Beighto : Aastra is pretty decent as well. |
| 21:53.08 | mog | RoyK, other than just now no ^_^ |
| 21:53.08 | [hC] | [TK]D-Fender: seriously? I hate those phones. |
| 21:53.15 | mog | i know benjk switched or is switching |
| 21:53.20 | mog | other than him no |
| 21:53.20 | [hC] | [TK]D-Fender: I have a bunch of 480i's and they 'work' but they are definitely not great. |
| 21:53.36 | robin_sz | sigh ... chan_zap.c:10734 setup_zap: Unknown signalling method 'bri_cpe' |
| 21:53.36 | [TK]D-Fender | [hC] : I learned how to provision them and they have a lot or really strong features that takes advantage of how * works as well. |
| 21:53.51 | [hC] | [TK]D-Fender: ah. my biggest beef was audio quality |
| 21:53.55 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:54.00 | [TK]D-Fender | [hC] : I'd still personally choose a Polycom over them, but depends on budget and deployement. |
| 21:54.11 | robin_sz | do I need to patch * for BRI capabability in zap ?? |
| 21:54.15 | Beighto | I noticed Polycom now has HD audio. Any word on this? |
| 21:54.38 | [TK]D-Fender | [hC] : Aastra's soft keys ARE amazing, but I do dislike the call handling more sepcifcally. Thats some that Polycom has over everybody period. |
| 21:54.58 | [TK]D-Fender | Beighto : HD = complete waste of time. the 650 isn't worth it now. |
| 21:55.15 | MstlyHrmls | what's the knock against the 650? |
| 21:55.32 | [TK]D-Fender | Beighto : mid/high end users = IP 501 / 480i, receptionist = IP 601 + Attendant modules. |
| 21:55.48 | [TK]D-Fender | MstlyHrmls : Its a great phone. Just comlpetely not worth the MONEY. |
| 21:55.59 | Beighto | thats too bad, sounded like a good idea. 430 has half duplex... too cheap? |
| 21:56.02 | [TK]D-Fender | MstlyHrmls : You don't get enought to warrant the cost increase |
| 21:56.17 | [TK]D-Fender | 430 = full-duplex + PoE. Great little phone |
| 21:56.23 | TexasJay | Hmm. I have a bunch of 480i's. I like 'em, but then again I'm a VoIP newbie and dont' know any better :-/ |
| 21:57.38 | MstlyHrmls | [TK]D-Fender: my guess is the 650 will start to shine when the high CPU load stuff (like SRTP) start rolling out |
| 21:57.52 | [hC] | i have an interesting array on my desk at the moment... a cisco 7970, 7941, polycom 601/430, and aastra 480i |
| 21:58.12 | MstlyHrmls | plus, it's got twice the flash and twice the RAM as the 60x |
| 21:59.27 | [TK]D-Fender | MstlyHrmls : posibly. |
| 21:59.29 | Beighto | [hC] so which is your favorite? |
| 21:59.56 | MstlyHrmls | [TK]D-Fender: but right now, it's a 601 with a big speaker and a backlight :-) |
| 22:00.14 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:00.35 | [hC] | Beighto: depends what im going for... :P and my opinion has been changing alot lately. the 7970 impresses me the most.. i have been leaning heavily towards polycom, but recently it seems as though ive found that the audio on a polycom handset is much more 'muffled' than a cisco. people tell me when i switch back and forth that the cisco is much clearer. |
| 22:01.13 | [hC] | Beighto: I like cisco alot, but i hate what it takes to sell them. They are predictable and reliable. Polycom is about 90% of the way there, and they have a much better price tag, too. |
| 22:01.31 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 22:03.19 | Beighto | maybe i'll have to buy aastra polycom and cisco |
| 22:04.04 | [hC] | Im also a fan of the linksys spa 922 and 942 |
| 22:07.51 | robin_sz | oh well, thats zaptel configured ... I suppose I might be able to compile the bristuff in a day or two |
| 22:07.56 | [TK]D-Fender | Cisco = waste, and just forget the Linksys stuff.... Polycom or Aastra both KILL them. |
| 22:08.13 | [TK]D-Fender | ok, off to martial arts, back much later.... |
| 22:08.20 | carrar | What about for BLF? |
| 22:08.31 | robin_sz | [TK]D-Fender: I have to agrre, that linksys SPA I just got is icky |
| 22:08.34 | [TK]D-Fender | For BLF : Polycom IP601 + Attendant modules. |
| 22:08.46 | carrar | I'll have to check those out |
| 22:08.59 | [TK]D-Fender | Linksys is CHEAP, but its screen usage blows as does it call handling capabilities. |
| 22:09.20 | robin_sz | you pays your money ... |
| 22:09.29 | [TK]D-Fender | I'm glad I ditched mine. The IP 301 feels nicer for crying out loud.... |
| 22:09.45 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:10.18 | [TK]D-Fender | I have an IP 301, 430, and 501 already, and I'm likly to pick up a 480i before long. I'm unsure about buying a 601 for personal use as it is a bit pricy |
| 22:10.31 | [TK]D-Fender | (for me at home), and I work on them all day at work. |
| 22:10.43 | [TK]D-Fender | I think a few Aastra's are in order for "well roundedness" |
| 22:10.45 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 22:10.47 | [TK]D-Fender | ok, BBIAB |
| 22:11.08 | iceyp | hey guys, i'm trying to get my pbx to recognize my phone number and if it's me calling to forward my call to another extension, is this possible? |
| 22:11.26 | iceyp | ohh and to add to it, I dont want the pbx to answer the call till the destination answers |
| 22:12.07 | iceyp | so ... my cell phone calls my pabx, my pabx notices my number and continues to ring without answering but forwards my call to another destination |
| 22:12.57 | iceyp | i tried adding my cellphone number to sip.conf and now the call just dies with the following: Found user '0274909712' Scheduling destruction of call 'C6FC83E6-53F411DB-A299C909-E573FDE8@202.180.73.144' in 15000 ms |
| 22:13.03 | iceyp | so i get 3 rings and then dead |
| 22:13.13 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:13.38 | iceyp | and I get a message from my provider saying 'sorry the call could not proceed at this time, please try again' which is like a congestion message |
| 22:14.08 | robin_sz | sigh .. would it not be simpler just to put all this BRI stuff into * as standard .. rather than these dangerous looking patches? |
| 22:15.14 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:18.13 | *** join/#asterisk icepack (n=gman0001@207.190.248.178) |
| 22:20.01 | *** join/#asterisk razu__ (n=razu@87-119-182-130.tll.elisa.ee) |
| 22:20.36 | icepack | i want to order dell 1950 and install TE212p on it, what pci bus should i order pci-x or pcie |
| 22:21.27 | Hymie | hmm |
| 22:21.29 | *** part/#asterisk Hymie (i=hymie@L8R.net) |
| 22:21.30 | *** join/#asterisk Hymie (i=hymie@L8R.net) |
| 22:21.48 | Hymie | does anyone know of a way to allow sip forwarding most of the time, but prevent it in a dial plan at certain times? |
| 22:22.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 22:22.18 | Hymie | I have an extension that dials three numbers, and if one of the SIP phones forward or have DND on, then of course bad things happen in general |
| 22:24.29 | razu__ | if i understand you right then you cant override phones DND and call forward ... you should configure the phone not to allow these functions |
| 22:25.17 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 22:25.47 | Hymie | razu__: and I have, but it would be nice to have those functions, plus to be able to set an option in the dial statement, such as "ignore call forwarding and dnd information" |
| 22:26.15 | Hymie | razu__: would be handy with SIP/100&SIP/102&SIP/104 if one person does a DND |
| 22:26.39 | razu__ | why does queue application act like this: If i'm the first caller in queue, I won't get the position announced, but if i'm second or third, then I get the announce of my position ... is there any flag for the queue to turn announcing on for the first caller? |
| 22:27.12 | Hymie | razu__: hmm, I get.. what it is... "you are next in line" or soemthing, sec |
| 22:28.07 | razu__ | Hymie : actually DND shouldn't do anything bad ... the call should still be placed for the others who'se line is free ... (as I remember) |
| 22:29.05 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:30.10 | E-Rage | Is there a way to have the internationalprefix= in Zapata.conf appaended to a call before it is passed into the dialplan? I seem to see posting from people claiming this is possible. |
| 22:30.15 | C6Vette | when doing a show queue whatever... How can i tell hold long its been since the received a call. It shows a (last was 30 secs ago) but |
| 22:30.22 | C6Vette | that since they hungup for the last call |
| 22:30.34 | Hymie | razu__: queue for me says "You are now first in line...." |
| 22:30.47 | razu__ | Hymie : I have this kind of conf right now ... http://razu.pri.ee/queue.txt ... but it doesn't work somehow :( |
| 22:31.13 | Hymie | razu__: I think it's a bug, but I have to do more research |
| 22:31.16 | Hymie | sec |
| 22:31.36 | Hymie | your config file is too long and complex, I can't understand it :P |
| 22:31.45 | razu__ | :) |
| 22:31.46 | *** join/#asterisk gennak0001 (n=Miranda@207.190.248.178) |
| 22:31.54 | razu__ | actually its quite simple |
| 22:31.56 | Hymie | you have periodic announce off, let me look in my config |
| 22:32.36 | Hymie | announce-frequency = 90 |
| 22:32.41 | Hymie | announce-holdtime = yes |
| 22:32.53 | Hymie | those two pop out at me |
| 22:33.23 | razu__ | I'm using asterisk 1.2.7.1 |
| 22:33.29 | razu__ | maybe it really has a bug |
| 22:33.31 | Hymie | about the same here |
| 22:33.33 | Hymie | could be |
| 22:33.41 | Hymie | Asterisk 1.2.4 |
| 22:33.43 | razu__ | but yours work fine ? |
| 22:33.51 | razu__ | for the first caller ? |
| 22:33.53 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:33.56 | Hymie | yeah |
| 22:34.05 | Hymie | it says "You are now first in line" after a bit |
| 22:34.15 | razu__ | hmm |
| 22:37.23 | Hymie | razu__: and SIP call forward is evil, it loses all context with the current call :/ |
| 22:37.28 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 22:38.17 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:38.58 | razu__ | Hymie : 1 question about the queue more ... does it announce on join or after playing some of the musiconhold ? |
| 22:39.18 | *** join/#asterisk postel_ (n=jp@wikimedia/Postel) |
| 22:40.57 | Hymie | it plays some of the music on hold irst |
| 22:40.59 | Hymie | first |
| 22:41.14 | teknoprep | where can i download the metermaid patch ? |
| 22:41.23 | razu__ | Hymie : about call forwarding ... yes the forwarding kicks ass :) I have bunch of pain with call forwarding, rdnis values etc ... I'm actually creating and planning special filters and routings just to get forwarding right :) |
| 22:42.06 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 22:42.48 | *** join/#asterisk r_evolution (n=no@208.251.203.5) |
| 22:42.52 | r_evolution | SUP SUCKAHS! |
| 22:42.52 | razu__ | Hymie : if you need multiple phones to ring at the same time ... i suggest you use the queue mechanism ... then you'll have better control over the calls |
| 22:42.57 | *** part/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 22:43.11 | r_evolution | anyone im interested in around? |
| 22:43.55 | Hymie | razu__: yeah, I guess. :/ But, I only want those phones to ring once each, for 30 seconds each, so.. a bit of a PITA :/ |
| 22:44.13 | r_evolution | hymie... like hymen :) |
| 22:44.18 | Hymie | and, anyhow... I bet $5 SIP forward and SIP DND would cause me pain.. PAIN I TELLS YA |
| 22:44.54 | r_evolution | if it makes you feel any better hymie... trying to make this ATA use IAX is causing me pain |
| 22:45.07 | justinu|laptop | give up on iax, man |
| 22:45.10 | r_evolution | it works beautifully for incoming IAX2 calls... just not so much for outgoing. |
| 22:45.16 | r_evolution | haha why justin? good to see you're around btw |
| 22:45.23 | razu__ | Hymie : actually I rember there was a flag about call forwarding max hops ... maybe if you set it to 1, then these forwards will be ignored ... thats how it should be as far as I know |
| 22:45.29 | justinu|laptop | you're not using iaxy are you? |
| 22:45.37 | r_evolution | fuck no |
| 22:45.41 | *** join/#asterisk liquidno2 (n=foo@cpe-68-201-147-69.sw.res.rr.com) |
| 22:45.46 | justinu|laptop | what other ATA speaks iax? |
| 22:45.48 | r_evolution | i have an ATA from Taiwan that supports IAX |
| 22:45.49 | r_evolution | :) |
| 22:45.52 | justinu|laptop | heh |
| 22:45.58 | r_evolution | you know I get fun toys |
| 22:46.00 | justinu|laptop | apparently not all that well :) |
| 22:46.01 | r_evolution | it's ORANGE! |
| 22:46.02 | Hymie | razu__: I didn't see it in the DIAL command wiki page, any idea on it? |
| 22:46.03 | r_evolution | no kidding ;x |
| 22:46.06 | r_evolution | well here's the hting |
| 22:46.11 | r_evolution | i've never done ANYTHING with IAX |
| 22:46.19 | r_evolution | so i'm not sure if the ATA is at fault |
| 22:46.20 | r_evolution | or if I am |
| 22:46.26 | r_evolution | b/c incoming calls are working beautifully |
| 22:46.26 | razu__ | Hymie : just a sec ... i'll try to find info about it |
| 22:46.30 | r_evolution | beautifully i tells you |
| 22:46.35 | justinu|laptop | try iax debug from the CLI |
| 22:46.36 | *** join/#asterisk Dude34 (n=Aces1UP@209.101.89.82) |
| 22:46.38 | r_evolution | OH! I've a WIFI phone from them as well |
| 22:46.44 | justinu|laptop | see if you can divine anything from those messages |
| 22:46.50 | r_evolution | I did... the ATA doesn't even appear to be passing the call to * |
| 22:47.01 | Dude34 | is there anyone here that runs a termination business that i could pick your brain for a few seconds? |
| 22:47.03 | liquidno2 | I have a voicemail configuration question. I have my voicemail line set to exten => 1,3,Voicemail |
| 22:47.04 | justinu|laptop | so no IAX packets on the server when you try and make a call |
| 22:47.14 | liquidno2 | damn it |
| 22:47.17 | liquidno2 | hit enter too soon |
| 22:47.23 | r_evolution | so it seems... |
| 22:47.25 | justinu|laptop | how about a local packet capture near the ATA, can you see it sending stuff? |
| 22:47.51 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 22:47.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:47.53 | liquidno2 | I have a voicemail configuration question. I have my voicemail line set to exten => 1,3,Voicemail(su1000), but when a call comes in, I get an error about no su1000 in the voicemail configuration |
| 22:48.20 | r_evolution | nah it's hooked into a switch and i dont feel like digging up a hub and setting up something to capture |
| 22:48.27 | Renacor | is there a way to specify an extension to the originate command? |
| 22:48.29 | r_evolution | see now the thing is... like i said... incoming it's working great |
| 22:48.47 | r_evolution | so I find myself wondering if maybe for some retarded reason the ATA isn't MEANT to use IAX on outgoing... maybe only incoming? |
| 22:48.55 | r_evolution | wouldnt that be retarded... but nothing surprises me anymore |
| 22:48.55 | justinu|laptop | can't imagine that |
| 22:49.12 | r_evolution | yeah well |
| 22:49.15 | justinu|laptop | anyways, without doing packet captures, you won't find out much about whats going on |
| 22:49.27 | r_evolution | i had a level 1 tech here ask me to help her find hte number for AOL |
| 22:49.33 | r_evolution | and i would've NEVER imagined that either |
| 22:49.36 | r_evolution | but it happened |
| 22:49.40 | r_evolution | so i dont put much of anything past people |
| 22:50.29 | justinu|laptop | you ever play around with ableton live? interesting music app |
| 22:50.53 | gennak0001 | is any irc channels for hardware compatability with Digium boards? |
| 22:50.54 | r_evolution | not me but my friend has |
| 22:51.01 | r_evolution | i dont do much for music production |
| 22:51.08 | Beighto | anybody use any fax software for asterisk? How is asterfax? |
| 22:51.09 | r_evolution | i've a few friends who produce dnb and progressive |
| 22:51.28 | *** join/#asterisk StyleWarz (i=stylewar@euphoria.evil-packet.org) |
| 22:51.36 | StyleWarz | Hello |
| 22:51.46 | StyleWarz | Where can i drop a feature request for zaptel/zaphfc? |
| 22:52.04 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 22:52.18 | fiber0pti | Does anyone have some free time to test out a new oeprator panel? |
| 22:52.18 | *** join/#asterisk alkiser (n=email@bdsl.66.14.163.224.gte.net) |
| 22:52.30 | StyleWarz | fiber0pti: for asterisk? |
| 22:52.37 | fiber0pti | correct |
| 22:52.47 | StyleWarz | i have a testing-asterisk box :) |
| 22:52.59 | fiber0pti | nice.. want to install my server and client? |
| 22:53.02 | justinu|laptop | i've love to sit down with an ableton expert and watch them make stuff |
| 22:53.03 | fiber0pti | it runs on java |
| 22:53.27 | StyleWarz | uh |
| 22:53.36 | StyleWarz | so i need jre and stuff for itß |
| 22:53.38 | fiber0pti | We want to test it before we go to general release |
| 22:53.43 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:53.46 | fiber0pti | Yes, you'll need JRE |
| 22:53.49 | fiber0pti | for server and client |
| 22:53.58 | StyleWarz | Naaah =) then i don't :) |
| 22:54.25 | fiber0pti | What operator panel do you use? |
| 22:54.33 | *** join/#asterisk alkiser (n=support@bdsl.66.14.163.224.gte.net) |
| 22:54.33 | StyleWarz | vim |
| 22:54.33 | StyleWarz | ;) |
| 22:54.41 | teknoprep | where can i get the metermaid patch for 1.2 ? |
| 22:54.52 | justinu|laptop | fiber0pti: which gui toolkit does it use? |
| 22:54.57 | r_evolution | hah... it's not as interesting as you might think justin |
| 22:54.57 | fiber0pti | SWT |
| 22:55.19 | justinu|laptop | apparently some DJs are using ableton for live sets |
| 22:56.31 | r_evolution | yep. |
| 22:56.35 | r_evolution | well lots of things actually |
| 22:56.38 | r_evolution | abelton is one |
| 22:56.51 | r_evolution | people also commonly use finalscratch... |
| 22:57.01 | r_evolution | or a herc board and abelton to simulate the pioneer CDJs |
| 22:57.30 | heison | <PROTECTED> |
| 22:57.30 | heison | Notice: Configuration file is /etc/zaptel.conf |
| 22:57.31 | heison | line 0: Unable to open master device '/dev/zap/ctl' |
| 22:57.33 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 22:57.57 | alkiser | is there anything i have to include in order to get MeetMe to work in 1.4... im getting an error when i call the extension for the room |
| 22:58.01 | alkiser | No application 'MeetMe' for extension |
| 22:58.14 | heison | i can't see zaptel in lsmod |
| 22:58.32 | heison | when tried to modprobe by hand, i get unresolvable symbol |
| 22:58.37 | heison | yet it compile just fine |
| 22:58.53 | heison | i'm upgrading to 1.2 |
| 22:58.54 | r_evolution | the thing about this ATA is that it sends SIP calls |
| 22:58.56 | r_evolution | but not IAX |
| 22:59.01 | r_evolution | so im like... wtf? |
| 22:59.22 | r_evolution | but it receives IAX O_o and SIP of course |
| 22:59.23 | r_evolution | ugh |
| 22:59.25 | r_evolution | confusing new shite. |
| 22:59.27 | justinu|laptop | weird |
| 22:59.32 | razu__ | Hymie : damn ... sorry I can't find any info about it ... maybe i read it from sip proto spec or smthing and * doesn't support changing that value |
| 22:59.34 | r_evolution | no kidding. |
| 22:59.43 | r_evolution | i've already emailed the techs with the company who make it |
| 22:59.51 | Hymie | razu__: doh :( thanks though |
| 23:02.55 | fiber0pti | Do you guys currently use an operator panel for asterisk? |
| 23:03.09 | r_evolution | OH MY FUCING GOD |
| 23:03.10 | r_evolution | justin... |
| 23:03.22 | r_evolution | justin please... please come sedate me before i fucking kill this dumb cunt. |
| 23:03.34 | justinu|laptop | lol |
| 23:03.37 | aptura | we dont talk like that here |
| 23:03.44 | r_evolution | who is this we? |
| 23:03.49 | liquidno2 | mouse in the pocket |
| 23:04.00 | r_evolution | I say what I want... |
| 23:04.50 | *** join/#asterisk RoyKa (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 23:04.52 | aptura | If you had a good attitude you would be working for a employer now ;) |
| 23:04.59 | r_evolution | hahahahahahahahaha |
| 23:05.02 | justinu|laptop | lol |
| 23:05.08 | r_evolution | there's the great irony :) |
| 23:05.15 | r_evolution | i AM working for an employer aptura... |
| 23:05.42 | mtgh | r_evolution: you sound like a 5 year old....most people find those words offensive and you might want to think before using them in a public fourm....but you are right, you are free to say what you want |
| 23:05.46 | justinu|laptop | this guy must be someone from the ministry for protection of virtue and prevention of vice |
| 23:06.09 | liquidno2 | My boss is an idiot... but then I work for myself... so... |
| 23:06.10 | r_evolution | mtgh... swearing is actually a normal and healthy part of the human emotional process |
| 23:06.16 | alkiser | is there anything special needed to get simple conferencing to work in 1.4, a module i have to enable etc? having no luck with it... |
| 23:06.26 | r_evolution | so basically... in a phrase. fuck off :) |
| 23:06.41 | Qwell | alkiser: install zaptel, install asterisk, modprobe ztdummy |
| 23:06.44 | Adam12 | r_evolution: And taking a crap is a part of the human digestive process but you dont' hear us talking about our bowel movements! |
| 23:06.58 | r_evolution | now... some people DO find them offensive... and that being the case... I will happily tone myself down... but not when 'TOLD' to |
| 23:07.06 | StyleWarz | Qwell: Are you a developer of zaptel? ;) |
| 23:07.15 | Qwell | StyleWarz: I can be |
| 23:07.21 | C6Vette | r_evolution dont expect much help from people when your talking like a 5 year old. |
| 23:07.26 | r_evolution | actually Adam many people do discuss issues with their system... especially when they have a problem |
| 23:07.36 | r_evolution | i don't expect help from most people in here anyway :) |
| 23:07.39 | StyleWarz | Qwell: I would give a feature request ;) Can somebody please implement Call Rerouting for point to point links? ;) |
| 23:07.51 | Qwell | StyleWarz: asterisk-dev mailing list |
| 23:08.06 | teknoprep | is there a patch for 1.2 to use MeterMaid ? |
| 23:08.09 | teknoprep | anyone ? |
| 23:08.23 | heison | for some reason, make install of zaptel doesn't copy all the .ko files into /lib/modules/2.x.x.x-686/extra |
| 23:08.37 | Qwell | heison: because your distro broke it |
| 23:08.40 | heison | so, zaptel and other modules failed to load |
| 23:08.57 | heison | i copied them by hand and ran depmod -a , it seems to work fine |
| 23:09.31 | heison | it works on the same distro, same version of the kernel on AMD Athlon, but not on my P4 box :( |
| 23:09.46 | r_evolution | so again... if you have an issue with the way I choose to express myself... I believe there is an X in the upper corner of your box... feel free to use it |
| 23:10.05 | justinu|laptop | there's also the ultra handy dandy ignore feature |
| 23:10.14 | r_evolution | precisely justin. |
| 23:10.34 | r_evolution | until such a time as someone who MATTERS asks that I tone it down |
| 23:10.42 | r_evolution | or the thought police come for me |
| 23:10.49 | Qwell | r_evolution: There is also the ultra useful kick button. :) There is a limit |
| 23:10.50 | r_evolution | i think i'll fuck and cunt and bitch all I want :) |
| 23:10.58 | r_evolution | lol |
| 23:11.02 | r_evolution | touche qwell ;) |
| 23:11.15 | r_evolution | hence the someone who matters :) |
| 23:11.29 | mtgh | Qwell: Thank you |
| 23:11.41 | Qwell | mtgh: I was playing Devils Advocate |
| 23:11.47 | r_evolution | hey mtgh... how bout use that handy ignore feature... or fuck off :) |
| 23:11.56 | r_evolution | qwell and i grew up together... |
| 23:11.59 | justinu|laptop | lol |
| 23:12.02 | r_evolution | he used to beat me up and take my lunch money :( |
| 23:12.03 | Qwell | riiiight |
| 23:12.18 | r_evolution | you still owe me a carton of milk... |
| 23:12.30 | r_evolution | you catch my earlier rant qwell? found an interesting little ATA from taiwan |
| 23:12.32 | r_evolution | does sip and iax... |
| 23:12.34 | Qwell | anyhow...I just finished driving 2,000 miles across the country, and unloading all my crap... |
| 23:12.45 | Qwell | time to relax and enjoy the AL rain |
| 23:12.46 | r_evolution | funny thing tho... it'll make an IAX call TO The ATA |
| 23:12.48 | r_evolution | but not from |
| 23:12.51 | r_evolution | 2,000 miles?!~ |
| 23:12.55 | r_evolution | roadtrip? |
| 23:13.10 | Qwell | r_evolution: moved to Huntsville |
| 23:13.18 | r_evolution | ahhhh... for a reason? or for fun? |
| 23:13.24 | Qwell | for Digium ;) |
| 23:13.29 | r_evolution | exactly |
| 23:13.35 | r_evolution | congrats then sir... as I assume you've been hired on |
| 23:13.39 | liquidno2 | he likes the mountain terrain |
| 23:13.49 | Qwell | r_evolution: for nearly 2 months now.. |
| 23:14.01 | r_evolution | eh I haven't been in here much man... I've been so swamped |
| 23:14.11 | r_evolution | we started running a calling card platform over * |
| 23:14.12 | tessier | Is it possible for two phones behind NAT send RTP direct to each other? I seem to recall it is not but can't remember why. |
| 23:14.13 | r_evolution | scary. |
| 23:14.20 | tessier | Behind different NATs I mean. |
| 23:14.22 | r_evolution | it's like 130K PINs... |
| 23:14.25 | tessier | With the SIP server on a public IP |
| 23:14.32 | r_evolution | LUCKILY... not all that many concurrent. |
| 23:14.33 | justinu|laptop | tessier: it's possible with STUN, but some trickery is needed. |
| 23:14.43 | justinu|laptop | tessier: check out how jingle (googletalk voice) does it |
| 23:14.46 | r_evolution | but I do like to push the limits. |
| 23:15.03 | tessier | justinu: Why will it not work without STUN? Both phones must then support STUN right? |
| 23:15.46 | justinu|laptop | because the phones need to figure out what IP address/UDP ports they're actually transmitting on after they pass thru the NAT |
| 23:16.03 | tessier | justinu: But asterisk can munge the SIP packets to work that out with nat=yes, right? |
| 23:16.24 | tessier | And it works for RTP from the phone behind NAT to asterisk. Why can it not do the same for both phones and send the RTP direct between the phones? |
| 23:16.38 | liquidno2 | What happened to reload in 1.4? |
| 23:16.52 | tessier | I know NAT screws everything up. Just trying to understand why there are workarounds for some cases but not so much for others. |
| 23:16.53 | justinu|laptop | yeah, but the two phones can't establish the bidirectional UDP stream to each other thru double NAT without first sending a stream to the other phone |
| 23:17.46 | justinu|laptop | and the far end NAT won't let it thru unless the far end phone establishes a UDP outbound on that port/ip combo |
| 23:17.47 | tessier | ah... |
| 23:18.56 | justinu|laptop | it depends on the kind of NAT you're behind too |
| 23:19.38 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 23:20.17 | tessier | ok, I got it now |
| 23:21.49 | *** part/#asterisk liquidno2 (n=foo@cpe-68-201-147-69.sw.res.rr.com) |
| 23:22.08 | r_evolution | so we're sitting around in the asterisk channel |
| 23:22.12 | r_evolution | what shall we do tonight? |
| 23:22.13 | r_evolution | oh wait |
| 23:22.14 | r_evolution | i KNOW |
| 23:22.22 | r_evolution | let's whine about someone using swear words O_O |
| 23:22.40 | tessier | What the fuck? |
| 23:22.42 | tessier | We can't swear? |
| 23:22.45 | r_evolution | hahah... |
| 23:22.59 | r_evolution | you know that's exactly what went through my head a bit ago tessier |
| 23:23.04 | justinu|laptop | yeah, i'd never heard that rule before now |
| 23:23.10 | justinu|laptop | this is irc |
| 23:23.12 | tessier | What next? No porn on the net? |
| 23:23.13 | r_evolution | someone told me that "we don't use that kind of l anguage" |
| 23:23.17 | marcus2 | so when will i be able to send called party info to the display of my polycom |
| 23:23.18 | justinu|laptop | if you want no swearing, go to AOL or something |
| 23:23.35 | r_evolution | that's what it was "we don't talk like that in here" |
| 23:24.20 | fiber0pti | Qwell: Would you like to try out a new operator panel? I'd like your feedback if you have some time |
| 23:24.33 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 23:24.40 | r_evolution | well is probably sleeping mang. |
| 23:25.06 | fiber0pti | o.. I just saw him talk.. owell.. |
| 23:27.22 | tessier | hmm....the reason I ask about NAT is that it would be nice if as much of our phone traffic as possible could go direct instead of having to go to our colo |
| 23:28.00 | tessier | If asterisk were smart enough to know that SIP packets coming from the same layer 3 ip but with different private IP's in the SIP packet don't need to be mangled but all others do that would be slick. |
| 23:36.27 | *** join/#asterisk Schulich (n=Jazba@165.154.103.197) |
| 23:36.44 | justinu|laptop | tessier: i don't think there's any reason why it couldn't be smart enough, it's just that no one has invested the time |
| 23:37.43 | justinu|laptop | tessier check the STUN RFCs |
| 23:38.02 | justinu|laptop | and also check out how jingle uses stun |
| 23:39.26 | teknoprep | exten => 71,hint,Local/71@parkedcalls |
| 23:39.34 | teknoprep | i have that in my context for using BLF on parked calls |
| 23:40.07 | teknoprep | its not working |
| 23:40.41 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 23:41.36 | *** join/#asterisk Kerry_G (n=Kerry_G@asterisk.geniusproducts.com) |
| 23:43.17 | Kerry_G | I'm setting up a dual span PRI card, have done plenty of single span, but with a dual span (46 B, 2 B) I am not exactly sure what the channel line in zapata.conf should be, should it be channel =>1-23 and channel=>25-47 or channel=>1-46? |
| 23:43.41 | *** join/#asterisk Ryushin (i=user@windwalker.openinnovations.com) |
| 23:44.01 | Ryushin | Is it possible to have multiple DID's coming to a analog line in the US? |
| 23:44.24 | Kerry_G | there are some carriers that do DID over POTS, but I dont think the hardware supports it |
| 23:44.35 | Ryushin | I'm building a asterisk box for my own small office, and I'm wondering weither I should get two analog lines, or a ISDN line. |
| 23:44.55 | Ryushin | ISDN BRI. |
| 23:44.58 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 23:45.35 | justinu|laptop | it's very uncommon, you might have a tough time getting DID info delivered on analog lines these days |
| 23:45.36 | Ryushin | I know ISDN can support DID's. I just don't know if Qwest can support it. I'm looking at the sangoma cards, so they would have to support DID's over analog? |
| 23:45.59 | justinu|laptop | i remember using dialogic DID/120 cards back in the day tho |
| 23:46.00 | Kerry_G | I know no carriers in California do it |
| 23:46.47 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 23:46.54 | *** join/#asterisk dovid (n=dovi5988@85.159.160.196) |
| 23:47.18 | teknoprep | can someone please help with BLF and pakred calls ? |
| 23:47.24 | teknoprep | i would like to monitor a parked call |
| 23:47.30 | Kerry_G | I dont know that you can |
| 23:47.55 | dovid | teknoprep: i dont think u can use BLF with parked calls |
| 23:47.59 | *** join/#asterisk justin_hur (i=r_evolut@208.251.203.246) |
| 23:48.05 | Ryushin | So I guess ISDN it is then. |
| 23:48.10 | teknoprep | i have read a few things that says you can |
| 23:48.11 | teknoprep | hmm |
| 23:48.24 | tessier | justinu: Yeah, looking into STUN now |
| 23:48.27 | teknoprep | http://www.voip-info.org/wiki/index.php?page=GXP-2000%20Extension%20Unit |
| 23:48.32 | teknoprep | that says you can |
| 23:48.33 | Ryushin | It would be nice if Sangoma would hurry up and finish their BRI card. They said that they are 3 months out from having it done. |
| 23:48.34 | teknoprep | hmmmmm |
| 23:49.57 | dovid | i guess u can monitor it |
| 23:49.58 | *** join/#asterisk docelmo (n=vircuser@m015f36d0.tmodns.net) |
| 23:50.09 | dovid | if u wana have diff buttons for diffrent parked cals |
| 23:50.11 | dovid | calls* |
| 23:51.42 | teknoprep | under show hints |
| 23:51.43 | teknoprep | -= Registered Asterisk Dial Plan Hints =- |
| 23:51.43 | teknoprep | <PROTECTED> |
| 23:51.48 | teknoprep | in the asterisk CLI |
| 23:53.29 | dovid | ok |
| 23:53.33 | dovid | u have a question ? |
| 23:53.55 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 23:54.23 | teknoprep | trying to make it work |
| 23:54.24 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 23:54.33 | teknoprep | been asking the same question for awhile |
| 23:54.38 | teknoprep | 701 is a Parking ext |
| 23:55.00 | teknoprep | trying to hint to ext 112 when a call is parked so BLF will light up on a button |
| 23:55.58 | Ryushin | What are some venders that everyone uses? I'm looking for Digium's BRI card, but telephonydepot doesn't carry it. |
| 23:56.33 | diablopico | can anyone tell me how to get the te212p to catch its oun interrupt on boot ? |
| 23:56.45 | file | the BRI card only works in Europe anyway |
| 23:56.59 | file | well, non-US/Canada |
| 23:57.29 | dovid | how about exten => 701,hint(Local/701@parkedcalls) |
| 23:57.34 | dovid | not working ? |
| 23:57.43 | dovid | wut kind of phone r u using ? |
| 23:57.54 | Ryushin | file: Well, it says it supports NT, which I need in the US. I'll have to do some more reading to make sure though. |
| 23:59.18 | raidenz | It seems the REALTIME app/function in the dialplan has changed from 1.2 -> 1.4. I used to have Realtime(sippeers|name|5000|Variable) but thats not valid in 1.4. Is the realtime function in 1.4 now *only* return one value or the array like in 1.2? I tried exten => 1,1,Set(MyVariable=${REALTIME(sippeers|name|1000)}) and nothing is returned. ( even tried viewing all variables with dumpchan). Any ideas? |