| 00:01.11 | pavlz | actually the initial project with GNUtoo-desktop was to speak about B.T.S. and how they work, |
| 00:02.31 | pavlz | for the fact the i want to do a sort of zine = magazine wrote in VIM, NANO, EMACS, i thinked to use as logo the GNU and to put the nick of GNUtoo-desktop |
| 00:03.25 | pavlz | on the other part i wrote GNUBTS, because i and GNUtoo-desktop are interested to speak about Free Networks |
| 00:03.39 | pavlz | Free as in Freedom |
| 00:04.35 | Naikrovek | mazpe: bootrom isn't called bootrom anymore. it's called updater or 'the updater' |
| 00:04.39 | Naikrovek | or something like that |
| 00:04.51 | Naikrovek | and it's version 5.something |
| 00:05.31 | pavlz | intially GNUtoo-desktop thinked to the OpenBTS, but here in Italy there are different problems |
| 00:05.48 | pavlz | for example the test licenses |
| 00:06.11 | jpsharp | Licensing spectrum is always the hard part. |
| 00:06.42 | pavlz | it's not clear if the Minister of Communication has or not and if can put to disposition the test licenses |
| 00:07.02 | pavlz | to pay a cheap amount to the year |
| 00:08.15 | pavlz | for example Chaos Computer Club has an OpenBTS, and German Minister of Communication gave them frequencies to work |
| 00:08.45 | pavlz | obviously they use OpenBTS for no-profit |
| 00:10.11 | pavlz | and mine interest is just no-profit, |
| 00:10.53 | pavlz | any of you was to the Chaos Communication Camp 2011 in FinowFurt ? |
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| 00:11.56 | pavlz | i was there, and i'll be in FinowFurt, next 2015 |
| 00:13.12 | pavlz | i invite to register yourself, few month before the 2015 CCC |
| 00:13.43 | pavlz | there was Emmanuel Goldstein of 2600.com |
| 00:13.55 | WIMPy | test licences in de are rather easy to get. |
| 00:14.40 | pavlz | there was ninux.org from italy, there was batman community and many other ones |
| 00:14.50 | pavlz | it was very interesting |
| 00:15.21 | pavlz | thank you WIMPy, i suppose to come to live near Berlin |
| 00:16.51 | WIMPy | Oh, and they are pretty good. I think it was for 20W at 6m. Quite a lot for testing. |
| 00:16.56 | pavlz | actually i have to go from the dentist, and i have problems with a pine in my garden, and i have to sell my house here in Roma, then i can left italy |
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| 00:18.30 | pavlz | if i don't solve before the problems with the pine, i could find in serious problems, and i'll forced to left italy |
| 00:18.42 | joobie | hey guys.. anyone got a script or a method used to stop/track calls sitting in a dial state? Had a situation where a call was made and the line just went dead.. i think it was sitting in the dial state but nthing on the other end.. ideally would like to either track that this is happening or send the call to another sip peer if it happens via the dialplan |
| 00:19.54 | pavlz | WIMPy are you a radioamateur ? "I think it was for 20W at 6m." |
| 00:20.09 | WIMPy | nope. |
| 00:20.53 | pavlz | what works to 20W at 6mt's ? |
| 00:21.10 | WIMPy | 6m height. |
| 00:21.29 | pavlz | you are speaking about the OpenBTS |
| 00:21.46 | WIMPy | GSM, yes. |
| 00:21.51 | pavlz | the fact is 20W are few |
| 00:22.53 | pavlz | to have a big network, it is necessary to have more ones, and GNUtoo-desktop told me that are not so much new |
| 00:25.26 | pavlz | i thinked that to contribute and help us is not important only to agree, but to write documentation to do a Free Network, and our interest is to put to disposition it, but asking a little contribute of 5 euros, helping us to buy instruments necessary to starts |
| 00:26.38 | pavlz | will be an initial announcement, and why we are starting to do a GNU Network which works with 100% Free Software |
| 00:26.52 | p3nguin | 6m isn't necessarily a favorite band of mine. |
| 00:27.48 | pavlz | and why we are interested to develop a Free Network to work with mobile phones |
| 00:28.36 | pavlz | infact, i don't think that OpenBTS works too on UMTS and wi-fi |
| 00:29.14 | pavlz | and i think that will be necessary to write much source code to be supported by OpenBTS |
| 00:30.14 | pavlz | i don't think that there is enough memory and a big cache for the UMTS and wi-fi traffic |
| 00:30.59 | pavlz | i suppose that will be necessary to split on different cells the traffic of the OpenBST goes down |
| 00:31.36 | pavlz | or the OpenBTS goes down |
| 00:34.17 | pavlz | there is much to work, and many to write, starting from the TELCO's and how these ones generate the signal which gives to the user the possibility to phone |
| 00:35.01 | pavlz | and all about the connection wireless between TELCO's and BTS, and all the traffic logged and trasmitted to the company |
| 00:35.35 | pavlz | how are mounted the antennas, cable used, connectors and much other things |
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| 00:37.00 | pavlz | who want join us, can start to contact me and GNUtoo-desktop and can writes in private or for e-mail to me to: hackers_space_italy@riseup.net |
| 00:37.10 | pavlz | any is free to join us |
| 00:37.26 | pavlz | thanks to you |
| 00:37.53 | pavlz | thanks to WIMPy, penguin.... |
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| 01:33.03 | jaytee | I have an Asterisk server running 1.8.11.1 and 18 Polycom 331 phones with SIP ver 3.2.4B. The server is behind a static nat on a Cisco ASA 5505. Inbound calling works but whenever someone attempts to call outbound the first call has no ringing and when the called party answers there is no audio or one-way audio. Calling a second time sometimes gives the same result but usually the third call |
| 01:33.04 | jaytee | attempt will produce ringing and 2 way audio. This also happens on internal extension to extension calls. SIP inspection is turned off on the Cisco. I've found several articles and most regarding Asterisk say to disable sip inspection. Since this problem affects internal phone to internal phone also I'm suspecting it might be a firmware issue with the Polycoms but I haven't found anything on |
| 01:33.04 | jaytee | their site to verify this. The configs I'm using I've used elsewhere and work fine except on this client's network. Anyone care to suggest what might be the cause? |
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| 01:59.10 | carrar | jaytee, might check to see if the ASA is doing any protoco fixup on your traffic |
| 01:59.15 | carrar | protocol |
| 01:59.54 | carrar | not sure if thats the same as 'inspection' |
| 02:00.47 | carrar | but if the problem happens from internal phone to another internal phone not passing through the ASA then it's obviosuly not the ASA |
| 02:01.56 | carrar | Try two xlite extensions |
| 02:02.07 | carrar | Try two internal xlite extensions |
| 02:02.25 | carrar | same subnet as asterisk |
| 02:02.34 | jaytee | good idea. I was also going to try two Polycoms with earlier firmware. |
| 02:02.36 | carrar | check the iptables |
| 02:03.10 | carrar | bbl (dinner time here) |
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| 05:46.37 | sawgood | is there a specific setting in /etc/asterisk/voicemail.conf to determine the 'mail' program used to deliver mail? |
| 05:46.56 | sawgood | I changed from sendmail to postfix, but sendmail keeps trying to send voicemails to email |
| 05:47.31 | kaldemar | mailcmd |
| 05:48.05 | sawgood | mailcmd ? |
| 05:48.54 | sawgood | I guess I don't understand (sorry) |
| 05:49.32 | kaldemar | sawgood: mailcmd is the name of the setting. |
| 05:49.35 | [TK]D-Fender | sawgood: /etc/asterisk/voicemail.conf <-- MAILCMD |
| 05:49.41 | sawgood | thank you! |
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| 05:50.52 | cchighman | hi |
| 05:52.04 | cchighman | Im using two 4-port digium cards connected to a dialogic tdm setup using isdn pri. dahdi shows ok on all 8 ports. however, pri show spans shows only the first 4 as being up. The other two pris on the second digium card are down. |
| 05:52.11 | cchighman | Any idea how to bring up these PRIs? |
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| 06:12.23 | urvg4 | hi all ,what is the extension context when using autocreatepeer in sip.conf? |
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| 06:14.20 | schmidts | good morning |
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| 06:41.04 | ChannelZ | aloha |
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| 07:45.52 | din3sh | mrning ppl |
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| 07:51.10 | cchighman | Im using two 4-port digium cards connected to a dialogic tdm setup using isdn pri. dahdi shows ok on all 8 ports. however, pri show spans shows only the first 4 as being up. The other two pris on the second digium card are down. |
| 07:51.17 | cchighman | Any idea how to bring up these PRIs? |
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| 09:16.29 | verywiseman | is it necessary to install astreisk-addons pkg to enable cdr_mysql in asterisk 1.8? |
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| 09:19.17 | ruben23 | hi guys where can i download free asterisk MOh..? |
| 09:22.21 | kaldemar | ruben23: if you install from source, you can select it to be downloaded upon install in the "make menuselect" menu. |
| 09:24.59 | kaldemar | verywiseman: addons are a part of the asterisk package in 1.8. cdr_mysql must be selected for install separately. |
| 09:25.20 | kaldemar | verywiseman: however you can also use cdr_odbc. |
| 09:25.49 | verywiseman | kaldemar, i tried to use cdr_odbc but it didn't work |
| 09:26.05 | verywiseman | would you see the configuration files? |
| 09:26.27 | kaldemar | ~ask |
| 09:26.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:27.12 | kaldemar | ruben23: also http://downloads.asterisk.org/pub/telephony/sounds/ has the packages. |
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| 10:42.59 | mahiti-irc | guys |
| 10:43.13 | mahiti-irc | i setting up chan_ss7 and asterisk |
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| 11:31.58 | youjelly | Is it possible to tell asterisk to send only specific events via the manager connection instead of all events? |
| 11:32.52 | WIMPy | manager.conf |
| 11:33.21 | WIMPy | Or you cann tell it per connection. Events is the keyword IIRC. |
| 11:34.13 | youjelly | When you make a connection to asterisk manager? |
| 11:34.49 | youjelly | in manager.conf, what I've read so far, there is an option to disable receiving all events not selective events |
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| 12:08.55 | gajini | Hi, I got the Error:- "" [ERR] ftmod_libpri.c:1317 -- Unable to get channel 1:-1" , while making call from ISDN phone |
| 12:11.10 | gajini | please help me to solve this issue "" [ERR] ftmod_libpri.c:1317 -- Unable to get channel 1:-1" , while making call from ISDN phone |
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| 12:26.39 | WIMPy | Never seen that. Any more information? |
| 12:29.09 | din3sh | never even used freeswitch |
| 12:31.10 | WIMPy | That's Freeswitch? |
| 12:32.03 | din3sh | seems so |
| 12:32.34 | din3sh | is it gajini? |
| 12:33.39 | WIMPy | wouldn't trust anythig using libpri. |
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| 12:34.42 | youjelly | is it possible to mute a single user on a conference call on asterisk, A,B,C,D are in a conf room and C doesn't want to hear what A is saying |
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| 12:35.42 | samfisher | Hi. I want to make my own PBX with internal numbers like 101, 102 etc. Each phone will be in different geographical locations and I only need internal voice service, no connection to PSTN. But, I want the phones to be able to call 911 and also if possible to send their location to 911. Is this possible? |
| 12:36.33 | leifmadsen | samfisher: for 911 you need interconnectivity for that with either a SIP provider, or you need to do all the research and connectivity yourself for 911 (non-trivial) |
| 12:36.44 | [TK]D-Fender | samfisher, 911 = PSTN. And this would mean you'd ahve to pick a connection per-device so that it could ID the location. |
| 12:36.55 | leifmadsen | samfisher: so I'd just get a credited account for like $20 or something and have that available for 911 services but not dial out services |
| 12:37.05 | WIMPy | youjelly: Only one participant for only one other participant? No. |
| 12:37.12 | leifmadsen | or get a single PSTN connection at each location (which is probably the safest) |
| 12:37.19 | leifmadsen | you could use your fax lines for that |
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| 12:38.07 | samfisher | leifmadsen: sound the most efficient solution |
| 12:38.09 | samfisher | thanks |
| 12:38.36 | samfisher | [TK]D-Fender: I can't afford a connection per device.. |
| 12:39.29 | [TK]D-Fender | samfisher, Go see what providers offer you as means of ID-ing the source then. There will probably be some sort of per-ID charge for this |
| 12:40.00 | [TK]D-Fender | samfisher, And I never said this would ahve to be a recurring charge.... PAYG instead. |
| 12:40.43 | WIMPy | And better check with both the ITSP and emergency services that this would actually work. |
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| 12:44.10 | leifmadsen | samfisher: you can typically only assign 1 address per DID, so you likely would need 1 DID per address/location |
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| 12:51.33 | samfisher | leifmadsen: DID mean phone number? searched on wiki but it's not very clear to me |
| 12:51.40 | samfisher | is it the same as ANI? |
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| 12:52.13 | WIMPy | When people say DID here, they usually mean directory number. |
| 12:53.04 | leifmadsen | ~did |
| 12:53.04 | infobot | somebody said did was Direct Inward Dialing, or just a phone number |
| 12:53.15 | WIMPy | ... wich are almost certainly non-DID |
| 12:53.31 | samfisher | directory number? |
| 12:54.07 | WIMPy | Phone number |
| 12:54.12 | *** join/#asterisk youjelly (~bwahahaha@182.177.62.248) |
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| 13:27.30 | Dovid | is there any way to build a specific module for asterisk with out going in to asterisk sources? i want to build res_snmp |
| 13:28.24 | leifmadsen | well if you build the module, what you're building it for likely won't like it anywqays |
| 13:28.43 | leifmadsen | it'll tell you it wasn't built with the same compile time options |
| 13:28.49 | leifmadsen | unless everything is exactly the same compile wise |
| 13:28.56 | leifmadsen | (which usually means you built it on the same machine) |
| 13:29.20 | Dovid | no i want to build on the same machine but it's a live box. dont want a restart. i just want to build the res_snmp module |
| 13:29.32 | tzafrir_laptop | Dovid, res_snmp is part of the asterisk source tree. building it outside the asterisk source tree would require for you to provide it the proper compile / link flags |
| 13:30.00 | leifmadsen | Dovid: so just build it per usual make process, copy file into module direct, then from CLI do module load res_snmp.so |
| 13:30.04 | leifmadsen | you don't need to restart asterisk |
| 13:30.08 | tzafrir_laptop | leifmadsen, you can get the compile-time options from /usr/include/asterisk.h or whatever (if installed therer) |
| 13:30.32 | Dovid | but it wont make by default. |
| 13:30.43 | leifmadsen | modify menuselect to make it build then |
| 13:31.30 | Dovid | ok. will try |
| 13:31.45 | leifmadsen | you're likely missing dependencies |
| 13:31.52 | Dovid | what snmp dpeendancies r needed |
| 13:31.53 | Dovid | ? |
| 13:32.00 | leifmadsen | not recommended to install on a production system unless you've already tested elsewhere |
| 13:32.09 | leifmadsen | menuselect will tell you what it needs |
| 13:32.19 | leifmadsen | also www.asteriskdocs.org |
| 13:32.19 | Dovid | leifmadsen: netsnmp? or doing this process? |
| 13:32.47 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:32.51 | Katty | helllllllllooooooooooooooooooooooooooooo nurse. |
| 13:32.56 | Dovid | sorry res_snmp |
| 13:33.07 | leifmadsen | Katty: +1 |
| 13:37.17 | *** part/#asterisk samfisher (~unaffilia@unaffiliated/samfisher) |
| 13:40.54 | Katty | it's going to be one of those itchy nose days. |
| 13:43.08 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:47.34 | Dovid | leifmadsen: I installed net-snmp and it still wont let me select from make menuselect. i did make install. do by any chance know what packages are needed from CentOS? |
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| 13:55.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:00.15 | alexscott | Hello there. Yesterday i have tried to use asterisk 10.3 as a t.38 gateway like scenarii from https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway. UC2 is working good but i want to be in UC1; how asterisk detectfax is working to initiate T.38 re-invite ? . UC3 isn't working at all after the t.38 re-invite from my ATA, my T.30 fax don't send any data; maybe something is wrong in establishment (preamble). Who has tried t.38 gateway with success ? |
| 14:01.53 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:07.05 | WIMPy | Dovid: Did you ./configure? |
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| 14:09.18 | puzzled | hi WIMPy |
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| 14:30.46 | WIMPy | hi puzzled |
| 14:32.07 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 14:37.01 | Dovid | WIMPy: Forgot that |
| 14:37.06 | ectospasm | alexscott: if you're receiving a fax, Asterisk should initiate the T.38 negotiation. Do you have t38pt_udptl = yes in sip.conf? |
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| 14:40.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:41.58 | alexscott | i have t38pt_udptl set to yes for my ATA peer (t38 side) but not set for the other peer (t30 side). |
| 15:00.21 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 15:01.45 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 15:02.44 | tzica | I have a question regarding VoIP and nat |
| 15:03.12 | tzica | I have a subnet 192.168.119.160/27 let's see with many restrictions |
| 15:03.48 | tzica | I have setup a VoIP client, give access to voip server - port5060 and I'm able to register |
| 15:05.12 | tzica | I'm able to call other extensions on others subnet but not able to hear each other |
| 15:08.58 | [TK]D-Fender | ~sipnat |
| 15:08.59 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 15:09.00 | [TK]D-Fender | ^^^ |
| 15:09.05 | alexscott | ~t38 |
| 15:09.05 | infobot | methinks t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 15:09.11 | [TK]D-Fender | Need more that just port forwarding |
| 15:09.23 | tzica | are there any access/ports that needs to be accepted between subnets ? |
| 15:10.02 | Faustov | asterisk 1.8.10.1, handset hang up during dial command results in this spammed 200 times to the CLI output: WARNING[9747]: app_dial.c:1379 wait_for_answer: Unable to write frametype: 2 |
| 15:10.05 | Faustov | any idea why? |
| 15:10.42 | jamko_ | tzica: generally 5060, 10000-20000, but there is much more to it, as stated by [TK]D-Fender. |
| 15:16.29 | *** part/#asterisk tzica (~wef@unaffiliated/tzica) |
| 15:16.39 | *** join/#asterisk tzica (~wef@unaffiliated/tzica) |
| 15:24.21 | leifmadsen | Faustov: because asterisk hadn't quite detected the hangup yet and was still attempting to send audio frames |
| 15:24.33 | leifmadsen | I get that sometimes, but usually only 2-3 times |
| 15:24.38 | leifmadsen | always after hangup |
| 15:24.43 | leifmadsen | nothing is *wrong* per se |
| 15:24.54 | Faustov | yeah, I was expecting one or two, but not so many |
| 15:25.25 | Faustov | for a moment I thought it's in a loop, but after 5 seconds it stopped |
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| 15:47.57 | *** join/#asterisk skirmisha (~vk@46.237.90.31) |
| 15:48.01 | skirmisha | hi guys |
| 15:48.07 | skirmisha | can someone help me on t38 |
| 15:49.23 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:50.51 | jeffspeff | How can I get the followme recordings located in /var/spool/asterisk to be in .ulaw instead of .sln? |
| 15:51.39 | Naikrovek | asterisk itself can convert them, I believe |
| 15:51.48 | Naikrovek | there's a cmdline option for that, iirc. |
| 15:51.54 | Naikrovek | I don't know what it is, though. |
| 15:52.10 | Naikrovek | also, sox may be able to do it, if it understands sln |
| 15:52.38 | Naikrovek | if you're wanting to change the recording format and not convert, I don't know. |
| 15:55.14 | coppice | skirmisha: asking a question is usually more productive than asking if you can ask a question |
| 15:57.18 | skirmisha | ok i got a fax issue with philips fax device |
| 15:57.25 | skirmisha | faxes are always smashed |
| 15:57.35 | skirmisha | when they are going over t38 using asterisk |
| 15:57.56 | skirmisha | so i am trying to figure out what cause that issue with smashes/crushed faxes |
| 15:58.10 | skirmisha | it is like half page ok and then just horizontal lines |
| 15:58.19 | skirmisha | not readable |
| 16:00.47 | coppice | is this something like philips fax<->ATA<->asterisk? |
| 16:04.01 | skirmisha | it is going over isdn, something like philips - isdn - asterisk - t38modem |
| 16:04.27 | skirmisha | there is patton in the middle, isdn - patton -asterisk |
| 16:07.15 | coppice | so you have spandsp acting as a T.38 gateway? |
| 16:08.16 | coppice | check that your ISDN card is deriving its clock from the PSTN, and not from an internal source. that is the usually cause of this kind of trouble |
| 16:09.52 | coppice | oh, the patton changes things. is the patton sending T.38 through asterisk to t38modem? |
| 16:09.53 | skirmisha | it is happening only to philips fax device |
| 16:10.01 | skirmisha | yes |
| 16:10.10 | skirmisha | patton is sending t38 |
| 16:10.30 | skirmisha | i am debugging 2 days now and can't get to anywhere |
| 16:10.43 | skirmisha | call is established over g711 and then switch over t38 |
| 16:11.02 | skirmisha | that part is fine , but only from that particular fax device faxes are crushed |
| 16:11.32 | skirmisha | i tested almost everything and now i am wondering whether it is signaling issue or data udptl |
| 16:11.36 | coppice | maybe the patton is not taking the clock for its ISDN interface from the PSTN |
| 16:11.49 | skirmisha | i can check that |
| 16:12.37 | skirmisha | on one of the port it is master |
| 16:12.42 | skirmisha | on the other it is slave |
| 16:12.50 | skirmisha | i do not think it is clock issue |
| 16:19.42 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:19.54 | c0rnoTa | Hi anyone |
| 16:20.16 | c0rnoTa | where I can find information about how to connect asterisk with madiant 2000 over MGCP ? |
| 16:20.26 | c0rnoTa | SS7 over MGCP |
| 16:20.46 | c0rnoTa | couldn't found any documentation in Internet |
| 16:21.10 | c0rnoTa | and no docs in distrib |
| 16:21.32 | c0rnoTa | any suggestions? |
| 16:22.25 | *** join/#asterisk bobman (~bobman@2001:470:7:3b::2) |
| 16:22.40 | WIMPy | How does MGCP and SS7 fit together? |
| 16:23.11 | Qwell | square hole, squarer peg |
| 16:23.38 | [TK]D-Fender | c0rnoTa, * does not support MGCP for anything other than phones. |
| 16:24.05 | [TK]D-Fender | So that would be "squarer peg being shoved in SIDEWAYS" |
| 16:24.29 | [TK]D-Fender | c0rnoTa, Re-flash <- |
| 16:27.05 | *** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net) |
| 16:28.13 | Nugget | yay, my D70 arrived |
| 16:28.21 | leifmadsen | neat |
| 16:28.29 | Nugget | how fancy |
| 16:29.43 | [TK]D-Fender | quite... |
| 16:30.21 | [TK]D-Fender | Looks like a neat toy. Interested in hearing how it measures up from your POV |
| 16:31.10 | leifmadsen | you fancy hun? |
| 16:31.16 | leifmadsen | s/hun/huh/ |
| 16:36.18 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:36.27 | c0rnoTa | WIMPy and [TK]D-Fender, as I understand task the question is to control signalling traffic of mediant 2000 over MGCP thru asterisk |
| 16:37.51 | WIMPy | So no ss7? |
| 16:39.02 | [TK]D-Fender | c0rnoTa, * only supports MGCP for phones. Not lines |
| 16:40.24 | c0rnoTa | scheme is: mediant with a lot of PRI interfaces connected to asterisk box over MGCP. And Asterisk has SIP peers. |
| 16:40.24 | c0rnoTa | Someone called from PSTN over PRI to my SIP peer. Asterisk saw incoming call to mediant over MGCP process it, throw call to SIP peer and then tell to mediant catch voice flow and transcode it from SIP peer directly. I hope that is not crooked explained :) |
| 16:40.38 | c0rnoTa | WIMPy: let's forget about SS7 ^) |
| 16:40.58 | c0rnoTa | [TK]D-Fender: Is there another channel, not MGCP for asterisk to do this? |
| 16:41.09 | *** join/#asterisk jacobkiers (~jjkiers@g172166.upc-g.chello.nl) |
| 16:41.16 | [TK]D-Fender | c0rnoTa, SIP. |
| 16:41.41 | [TK]D-Fender | [12:24} c0rnoTa, Re-flash <- |
| 16:42.30 | c0rnoTa | [TK]D-Fender: couldn't understand what's mean "RE-flash" :) |
| 16:44.38 | c0rnoTa | [TK]D-Fender: so, I can connect mediant with asterisk thru SIP and using options like directmedia pass voice flow directly to mediant, am I right? |
| 16:44.45 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:45.42 | [TK]D-Fender | Depends what the Mediant supports. |
| 16:45.56 | [TK]D-Fender | You should probably be reading its manual right now.... |
| 16:46.44 | jacobkiers | Hello, does anyone know how to forward/pass-through the SIP response code from one channel to another? I'm trying to dial forward (i.e. incoming call, exten => 1234,1,Dial(SIP/otherdestination@trunk)), but when the trunk sends me a 404 Not Found, the original channel (the one which dials into Asterisk), is sent a 480 instead of the expected 404. |
| 16:47.05 | Qwell | jacobkiers: Asterisk is not a SIP proxy |
| 16:47.57 | leifmadsen | it is a B2BUA, which means the channels are independent of each other |
| 16:48.04 | leifmadsen | Asterisk allows them to talk, through Asterisk |
| 16:48.17 | [TK]D-Fender | * matches the call so it is not a 404 |
| 16:48.50 | c0rnoTa | [TK]D-Fender: thanks for advice :) but I have no device too. In fact I'm only interesting in asterisk functionality. You already answered on my question - asterisk support only MGCP phones. |
| 16:49.02 | jacobkiers | So, it is not possible to read the result from the forwarded channel and feed it back to the original? |
| 16:49.28 | [TK]D-Fender | jacobkiers, it isn't "forwarded". Never ever call it that. |
| 16:49.38 | [TK]D-Fender | B2BUa <------- |
| 16:50.09 | jacobkiers | I tried with Hangup(${HANGUPCAUSE}), but that didn't work. |
| 16:50.10 | [TK]D-Fender | jacobkiers, "core show application hangup" <- |
| 16:50.44 | Katty | omnomnoms |
| 16:50.47 | WIMPy | suspects that the issue is that HANGUPCAUSe isn;t set to what you'd expect. |
| 16:51.29 | Katty | i'll expect your set in a minute. |
| 16:52.06 | [TK]D-Fender | Katty, My set plays later tonight ;) |
| 16:52.13 | c0rnoTa | [TK]D-Fender: thanks for support, but what means Re-flash? ))) |
| 16:52.16 | Katty | brown chicken brown cow |
| 16:52.16 | [TK]D-Fender | Katty, And recording due shortly... |
| 16:52.29 | [TK]D-Fender | c0rnoTa, replace the firmware on it <- |
| 16:53.38 | c0rnoTa | [TK]D-Fender: Oh, thanks! Right! I could not link it with topic |
| 16:57.18 | verywiseman | Is there web interface for CDR stored in mysql? |
| 16:57.32 | [TK]D-Fender | verywiseman, Plenty. Go check the Wiki fora list of them |
| 17:02.37 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:04.03 | *** join/#asterisk timahvo1 (~rogue@197.176.109.145) |
| 17:05.03 | jacobkiers | WIMPy: That is correct. The hangupcause 34. However, the responsecode from the trunk is 404 (SIP), so I would expect something in the 1-4 range |
| 17:11.27 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:13.30 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:21.51 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 17:22.27 | adeel|work | if i hit the dial timeout for answering a call, is there a way to change the SIP response code? |
| 17:29.37 | [TK]D-Fender | adeel|work, "core show application hangup" <- |
| 17:34.04 | adeel|work | [TK]D-Fender, ah, thanks...don't happen to know of a resource listing the hangup cause code mappings do you? |
| 17:35.09 | [TK]D-Fender | adeel|work, It's well lsited in the source |
| 17:35.32 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 17:38.19 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
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| 17:50.17 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:52.54 | jacobkiers | So, after some investigation I found out the following: immediately after the Dial statement, the HANGUPCAUSE is 34. A tcpdump shows the result from the trunk to be a 404 Not Found message. Did I do something wrong by using ${HANGUPCAUSE}, or didn't I correctly understand you all? |
| 17:54.37 | *** join/#asterisk GeoGeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 17:54.57 | GeoGeek | Hey...anybody got experience with multicast paging on Yealink phones? |
| 17:57.37 | GeoGeek | Sure is quiet in here today... |
| 17:58.26 | *** part/#asterisk GeoGeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 17:58.31 | *** join/#asterisk GeoGeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 18:03.12 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 18:03.54 | *** join/#asterisk GeoGeek_ (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 18:04.54 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:06.29 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 18:09.14 | GeoGeek | Hmm |
| 18:18.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:36.40 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:37.05 | atan | Anyone know a simple step by step guide on how to setup 10.4 using MySQL as the database for SIP devices? |
| 18:37.51 | atan | The details I have come across so far seem very outdated. They show mysql table layouts that are for like 1.4 I'd swear |
| 18:38.15 | [TK]D-Fender | ~book |
| 18:38.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:38.17 | [TK]D-Fender | ^ |
| 18:38.31 | leifmadsen | look at Relational Database chapter |
| 18:38.31 | atan | I had high hopes there might be something a tad smaller, no? |
| 18:38.36 | [TK]D-Fender | And there should also be sample table layouts in the tarball |
| 18:38.41 | leifmadsen | Relational Database Integration* |
| 18:38.52 | leifmadsen | atan: don't follow the entire chapter -- just read what you need |
| 18:39.10 | leifmadsen | most of the chapter is for hot-desking anyways, so just skip the example |
| 18:39.29 | leifmadsen | just read the sections related to setting up the database, tables, and realtime |
| 18:40.36 | atan | Okie dokie! :-) |
| 18:40.40 | bn-7bc | ok this is strange, I have the following in my voicemail.conf : 101 => 1234,Bjarne,bjarne-imp@holmedal.net,,Tz=Oslo,attach=yes,saycid=yes,dialout=out,callback=out,review=yes,operator=yes,envelope=yes,moveheard=yes,sayduration=yes,saydurationm=1 |
| 18:40.47 | atan | Thank you fellas, yet again |
| 18:41.59 | bn-7bc | but when i check the voicemail for that user it seems to ignore everything after TC= i.e. i don't here the cid etc |
| 18:42.39 | bn-7bc | ups that was Tz= |
| 18:42.46 | bn-7bc | any idfeas |
| 18:42.49 | *** join/#asterisk steve-o_ (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
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| 19:31.38 | *** join/#asterisk picard276 (~chatzilla@rrdhcp-122-13.redrover.cornell.edu) |
| 19:31.43 | picard276 | hey guys i had a question about VoIP with SMS |
| 19:31.57 | picard276 | so if i have a DID from a provider and i have a KAnnel server attachd to my asterisk box... can i do SMS messaging? |
| 19:32.05 | picard276 | or does my Voip provider need to support SMS? |
| 19:32.29 | *** join/#asterisk hypsium (c4d9d9c5@gateway/web/freenode/ip.196.217.217.197) |
| 19:33.37 | atan | leifmadsen, perhaps you might know how to define a server (other than localhost) in res_odbc.conf? I have it set in my /etc/odbc.ini and /etc/odbcinit.ini but Asterisk stills seems to try logging in to a server at localhost. I'm puzzled. |
| 19:35.03 | hypsium | hello, I need your help. Using debian, asterisk 1.8, dahdi 2.6 and a DIGIUM TC400B transcoder card (2007 REV A1) . Everything is ok, but when loading wctc4xxxp, i get failed to load firmware error -5. It's not firmware file not found, but something else and no help on Internet about this error. I found a bug declaration, closed in 2009, asking to see with Digium support. Anyone have any idea plz? |
| 19:38.31 | pabelanger | hypsium: IIRC debian does not distribute firmware files. I believe you need to install dahdi-firmware |
| 19:38.34 | leifmadsen | atan: just set Server value in /etc/odbc.ini |
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| 19:38.57 | leifmadsen | atan: and set the dsn value that matches what you've defined in odbc.ini |
| 19:39.08 | pabelanger | Which is in the non-free repo |
| 19:39.29 | leifmadsen | right, probably need to use the digium links |
| 19:39.41 | atan | leifmadsen, see this is the funny thing I have Server = set in there. It's ignoring it. |
| 19:39.46 | leifmadsen | I had to add links to the digium repos for firmware for CentOS |
| 19:39.51 | Qwell | leifmadsen: Do you know if you documented the cdr_odbc.conf in the book? |
| 19:39.51 | leifmadsen | atan: unknown then |
| 19:40.01 | leifmadsen | Qwell: I know that I did or did not document it |
| 19:40.14 | Qwell | I like those odds. |
| 19:40.36 | atan | leifmadsen once I modify that file must I reload anything else outside of Asterisk? |
| 19:40.57 | leifmadsen | unknown (probably not) |
| 19:41.33 | bluregard | does Ringing() not work on an answered channel? |
| 19:42.40 | pabelanger | what type of channel? |
| 19:42.49 | bluregard | sip |
| 19:42.57 | pabelanger | then no |
| 19:43.11 | pabelanger | don't answer it |
| 19:43.21 | pabelanger | and use Progress() |
| 19:44.30 | [TK]D-Fender | bluregard, It should work in the sense of actually generating tone in-band for an already answered call..... |
| 19:44.46 | bluregard | it doesn't |
| 19:44.57 | [TK]D-Fender | Progress() shouldn't impact an answered call I would imagine... |
| 19:45.11 | bluregard | if I comment out Answer() I get the ringing tone |
| 19:45.44 | [TK]D-Fender | bluregard, Playtones(ringing) should do it IIRC |
| 19:45.58 | bluregard | ahhh, let me try that |
| 19:46.11 | leifmadsen | bluregard: because asterisk can't send a 180 Ringing() after the answer |
| 19:49.34 | mjordan | nothing should send a provisional response after a final response. |
| 19:50.48 | bluregard | that works great |
| 19:51.38 | bluregard | asterisk answers the google voice call, sendDTMF(1) and then rings while the sip peer is dialed. |
| 19:52.05 | bluregard | [TK]D-Fender, and its playtones(ring) btw |
| 19:52.20 | [TK]D-Fender | 3 extra chars free! |
| 19:53.06 | bluregard | ;) |
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| 20:02.01 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:23.36 | p3nguin | If you use Ringing() followed by some wait time so it can actually produce ringing, it will make the ringing sound. |
| 20:24.07 | p3nguin | Try Ringing() followed by Wait(10) to see. |
| 20:25.11 | pabelanger | playing ringtone inband is evil |
| 20:25.13 | pabelanger | just saying |
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| 20:28.24 | geogeek | Having trouble with voicemails....when set to wav or wav49 I get this: check_header: Does not begin with RIFF |
| 20:28.53 | geogeek | Asterisk 1.8.11 |
| 20:29.56 | pabelanger | geogeek: solaris? |
| 20:31.08 | geogeek | pabelanger: No, I saw that bug...running on Ubuntu |
| 20:31.54 | pabelanger | I don't remember seeing anything related to format_wav changing recently |
| 20:32.02 | pabelanger | did this work in 1.8.10? |
| 20:32.05 | pabelanger | or do you know? |
| 20:32.59 | geogeek | pabelanger: I don't know...but it did work with 1.8.11...just isn't working now. Not sure what I may have done to break it. |
| 20:33.21 | geogeek | pabelanger: I tried gsm and g722 and they both give me zero-byte-lenght files. |
| 20:33.28 | geogeek | *length |
| 20:34.04 | pabelanger | Actually, I lie. They did some changes recently. |
| 20:34.14 | pabelanger | try 1.8.10 or sooner |
| 20:34.19 | pabelanger | might be a regression |
| 20:35.24 | geogeek | pabelanger: What about another format? Really not wanting to go backwards...it did work on 1.8.11 I am sure because that's the only release I have used since the freepbx. |
| 20:37.11 | pabelanger | You can try, but at the end of the day, finding out if something broke recently will help developer to fix the issue. And, if other people on 1.8.11 have the problem, then it is a regression. |
| 20:37.23 | pabelanger | You can try 1.8.12.0 too |
| 20:37.29 | pabelanger | which includes a few fixes |
| 20:37.36 | pabelanger | http://svnview.digium.com/svn/asterisk/branches/1.8/formats/format_wav.c?view=log |
| 20:37.40 | pabelanger | not sure if they are the issue |
| 20:37.40 | geogeek | When did that release? |
| 20:37.48 | Kobaz | see topic |
| 20:37.54 | Kobaz | <PROTECTED> |
| 20:38.03 | Kobaz | yesterday |
| 20:39.14 | geogeek | Oh yeah...there it is! |
| 20:40.19 | geogeek | Thanks. |
| 20:40.23 | atan | Well I'm lost. I've changed every server= and Server= and Server => and server => I can find in /etc/odbc.ini, /etc/asterisk/res_odbc.conf and yet I still get " WARNING[16937]: res_odbc.c:1552 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=1044 [unixODBC][MySQL][ODBC 5.1 Driver]Access denied for user 'asterisk'@'localhost'" |
| 20:40.58 | atan | The mysql server is not localhost, I've set it to a remote server in /etc/odbc.ini yet it's no-go = |
| 20:41.07 | geogeek | Now I have another question....does anyone have experience with multicast paging using Yealink phones? I get the phones to go into multicast mode but the speakers don't turn on. Yealink support has usually been responsive but not on this issue. |
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| 20:48.17 | atan | Perhaps this is a bug then. If I place Server = 192.168.1.1 or google.tk inside /etc/odbc.ini Asterisk shows an error about unable to resolve/connect to host, whatnot. If I put only a valid domain or IP it complains about connecting to the localhost. I must have missed something somewhere. |
| 20:49.12 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:52.28 | atan | Ahh snap. Got it. Silly config files :-) |
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| 23:02.12 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:02.12 | *** topic/#asterisk is #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.0 (2012/05/02), 1.8.12.0 (2012/05/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:03.04 | urvg4 | hi all,where do I find failed registration logs on asterisk? |
| 23:04.51 | hypsium | urvg4 for the basic configuration, i think in /var/log/message |
| 23:05.04 | hypsium | anyone about the mean of the -5 error value when wctc4xxp module starting and trying to load the firmware (dahdi-fw-tc400m.bin) for TC400B? |
| 23:11.33 | atan | I want to learn the mailbox var set for a SIP user using realtime. According to http://www.voip-info.org/wiki/view/Asterisk+func+sippeer it's not supported. Is there a way around this? Idea being I want the user to collect their voicemail by dialing out from their phone without needing to know their mailbox number :-) |
| 23:14.25 | din3sh | hypsium: where is your dahdi-fw-tc400m.bin located? |
| 23:14.40 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 23:14.56 | urvg4 | hypsium: checked and found no reg failures but still have a soft phone with unauthorized error? |
| 23:15.13 | hypsium | din3sh /lib/firmware and /usr/lib/hotplug/firmware/ , as the make install-firmware rule act. |
| 23:16.41 | hypsium | urvg4, check your log config on logger.conf |
| 23:16.58 | din3sh | does dmesg now show firmware loaded? |
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| 23:18.42 | hypsium | din3sh , no ... wctc4xxp 0000:00:09.0: firmware: requesting dahdi-fw-tc400m.bin -- wctc4xxp 0000:00:09.0: Failed to load firmware. -- wctc4xxp: probe of 0000:00:09.0 failed with error -5 |
| 23:19.19 | hypsium | if I put out /usr/lib/hotplug/firmware out, I get error -2 instead. |
| 23:19.19 | din3sh | for some reason |
| 23:19.39 | din3sh | the firmware isnt getting loaded properly |
| 23:19.52 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:4d6d:f0e6:a685:caf1) |
| 23:20.13 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 23:20.30 | urvg4 | hypsium found these: [logfiles] |
| 23:20.31 | urvg4 | #include logger_logfiles_additional.conf |
| 23:20.31 | urvg4 | #include logger_logfiles_custom.conf |
| 23:20.31 | urvg4 | console => notice,warning,error,debug |
| 23:20.31 | urvg4 | messages => notice,warning,error |
| 23:20.34 | hypsium | din3sh, yes, and I'm looking for the mean of the -5 error. Any ideas about source files I can check for investigate about the mean of this error? |
| 23:20.49 | *** part/#asterisk atan (~atan@unaffiliated/atan) |
| 23:21.38 | din3sh | http://downloads.digium.com/pub/telephony/firmware/releases/ |
| 23:21.49 | hypsium | din3sh, i found this : https://issues.asterisk.org/view.php?id=16417 - but how I can ask the support, the card was bought on internet, from a person.. |
| 23:21.50 | din3sh | try downloading it manually |
| 23:22.22 | hypsium | din3sh i download it manually and replace, modprobe -r wctc4xxp and reload the modules, same error. |
| 23:22.38 | din3sh | :/ |
| 23:22.58 | hypsium | that's why I was looking for old firmware, or may be I'm gonna try zaptel and 1.6 and try ... |
| 23:23.10 | hypsium | thanks for u interest and ur help |
| 23:23.46 | urvg4 | hypsium how do I get or set up failed reg logs? |
| 23:24.19 | din3sh | which card is that anyway? |
| 23:24.25 | urvg4 | do I add this full => notice,warning,error,verbose? |
| 23:26.00 | din3sh | add debug also |
| 23:26.05 | din3sh | am off to bed |
| 23:26.07 | din3sh | night |
| 23:26.17 | hypsium | din3sh TC400B digium |
| 23:26.49 | hypsium | dins3sh tx |
| 23:26.53 | hypsium | gn |
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| 23:53.47 | atan | Is it possible to reveal what passwords a connecting client is trying to use to connect? |
| 23:56.39 | ectospasm | not usually, auth sends a MD5 hash of the password, that may be salted... |
| 23:57.02 | *** part/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 23:57.05 | atan | Well crap then that idea's out. :-) |
| 23:57.59 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |