| 00:00.08 | *** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 00:01.13 | DerkKo | Hey guys.. i been trying to make asterisk 1.8 receive Faxes…. the documentation that can tell me how to accomplish receiving fax step by step |
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| 04:18.44 | fling | Hello. |
| 04:19.14 | fling | fulcan: can you help me please with sip setup? |
| 04:21.43 | fling | pbx ~ # nmap localhost -sU -p 5060 |
| 04:21.44 | fling | 5060/udp open|filtered sip |
| 04:21.55 | fling | same with sample config files… |
| 04:22.32 | fling | same with this > http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
| 04:25.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:31.07 | fling | If I remove all the config files but /etc/asterisk/sip.conf > https://gist.github.com/3031114 |
| 04:31.10 | fling | 5060/udp closed sip |
| 04:32.43 | fling | pbx ~ # netstat -anp --udp --tcp > https://gist.github.com/3031117 |
| 04:34.48 | ChannelZ | is asterisk even running? |
| 04:35.51 | fling | ChannelZ: pbx ~ # ps ax | grep asterisk > https://gist.github.com/3031123 |
| 04:40.00 | fling | is Asterisk 1.8.12.1 fine? I will try 10.5.1 |
| 04:42.48 | fling | asterisk it not listening! |
| 04:43.32 | ChannelZ | well if your netstat output was true, it's not listening to anything. |
| 04:43.56 | ChannelZ | you said you wiped out all the configs, that probably has a little to do with it |
| 04:46.29 | fling | ChannelZ: so what should I do? |
| 04:46.46 | ChannelZ | do you get an error if you do "module load chan_sip" ? |
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| 04:51.29 | fling | ChannelZ: no, asterisk started listening! http://dpaste.com/766249/ |
| 04:52.41 | fling | ChannelZ: 'require = chan_sip.so' > asterisk/modules.conf ? |
| 04:52.49 | ChannelZ | ok well you should start by putting back all your configs, or at minimum asterisk.conf, modules.conf, indications.conf, logger.conf, rtp.conf |
| 04:53.08 | ChannelZ | it should load automatically if your autoload is turned on |
| 04:55.00 | fling | ChannelZ: thanks! Is it safe to use these default config files? I want to use sip only |
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| 04:59.35 | fling | ChannelZ: looks like autoload is not working |
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| 05:55.41 | fling | ChannelZ: how to make chan_sip to load at start? |
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| 06:06.33 | fcrs | is there something similar to ${DB_EXISTS(blacklist/${CALLERID(num)})} that works with patterns such as 1234XXXXXXX |
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| 06:09.34 | fcrs | basically, what's a good way to build a blacklist that allows for pattern matching? |
| 06:22.16 | ChannelZ | fling: you probably have some other issue, look for errors on startup |
| 06:23.41 | fcrs | i think this will do what i want if i just put the first 3 digits in the database: ${DB_EXISTS(blacklist/${CALLERID(num)}:0:3)} |
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| 06:26.24 | v0lZy | lo |
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| 06:39.10 | fling | ChannelZ: Jul 2 13:38:45 pbx asterisk_wrapper: Initializing asterisk wrapper |
| 06:39.17 | fling | ChannelZ: nothing interesting in syslog |
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| 06:42.35 | ChannelZ | I mean from asterisk direct. Shut it down, run it manually with -cfvv or such, and see what it says from the get-go |
| 06:42.56 | ChannelZ | if autoload isn't working, and chan_sip isn't even loading on its own, something is jacked |
| 06:43.19 | fling | ChannelZ: it is loaded manually… |
| 06:44.36 | ChannelZ | <fling> ChannelZ: how to make chan_sip to load at start? |
| 06:45.01 | fling | ChannelZ: ok, https://gist.github.com/3031494 |
| 06:45.23 | *** part/#asterisk fcrs (~fcrs@unaffiliated/fcrs) |
| 06:46.31 | ChannelZ | looks like you have some syntax errors |
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| 06:49.36 | fling | ChannelZ: [Jul 2 13:44:23] ;31mERRORm[29161]: mnetsock2.cm:m263m mast_sockaddr_resolvem: getaddrinfo("dynamic ", "(null)", ...): Name or service not known |
| 06:50.50 | ChannelZ | Yes that's a puzzle as well |
| 06:53.33 | ChannelZ | and probably the same answer |
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| 06:58.37 | *** join/#asterisk gusto (~gusto@82.113.121.50) |
| 06:59.05 | fling | ChannelZ: https://gist.github.com/3031558 https://gist.github.com/3031545 [Jul 2 13:55:45] [1;33mNOTICE[0m[29641]: [1;37mloader.c[0m:[1;37m1129[0m [1;37mload_modules[0m: 1 modules will be loaded. |
| 06:59.12 | fling | SIP channel loading... |
| 07:00.08 | fling | but it is still not listening |
| 07:01.21 | fling | ChannelZ: listening now! |
| 07:01.31 | fling | ChannelZ: thanks… |
| 07:02.38 | gusto | what is going on? |
| 07:02.39 | fling | but only if I start it directly with asterisk -cfvv |
| 07:03.01 | gusto | what version are you using? |
| 07:03.16 | fling | gusto: asterisk is not listening if starteg with initscripn and sip.conf > https://gist.github.com/3031558 |
| 07:03.25 | fling | Asterisk 1.8.12.1 |
| 07:03.35 | gusto | what does netstat -aut say? |
| 07:05.34 | gusto | it should listen at UDP port 5060 per default |
| 07:05.55 | gusto | so netstat -lun should show something like that |
| 07:07.00 | gusto | did you upgrade or is it a fresh install of modules? |
| 07:07.13 | gusto | on what operating system are you there? |
| 07:07.30 | fling | Linux pbx 3.3.8-gentoo-gnu_fling #4 SMP Fri Jun 29 17:15:28 NOVT 2012 x86_64 Intel(R) Pentium(R) D CPU 2.80GHz GenuineIntel GNU/Linux |
| 07:07.33 | fling | no multilib |
| 07:07.51 | gusto | gentoo? |
| 07:08.20 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:08.21 | fling | gusto: yes, pbx ~ # netstat -aut > https://gist.github.com/3031611 |
| 07:08.21 | schmidts | good morning |
| 07:08.46 | gusto | gentoo are bullshitters, they deadpatch everything till it's broken |
| 07:08.47 | fling | gusto: it is listening if started with asterisk -cfvv |
| 07:09.16 | gusto | udp 0 0 *:5060 *:* |
| 07:09.25 | gusto | here you have that it's listening |
| 07:09.55 | gusto | when you start it as a service it does not listen? |
| 07:10.05 | fling | right |
| 07:10.05 | gusto | well, maybe there is something wrong with the modules then |
| 07:10.21 | gusto | maybe he does not load chan_sip |
| 07:10.33 | fling | yes, I can load sip manually and it starts listening |
| 07:10.41 | gusto | or the idiots from gentoo put something in that prevents it |
| 07:10.58 | fling | pbx ~ # ls /usr/portage/net-misc/asterisk/files/1.8.0/ |
| 07:10.59 | fling | asterisk.confd asterisk.initd asterisk.initd2 find_call_ids.sh find_call_sip_trace.sh |
| 07:11.05 | fling | gusto: no patches |
| 07:11.06 | gusto | well, then that idiots are blocking it in /etc/asterisk/modules.conf or something like that |
| 07:11.59 | fling | gusto: I have added preload = chan_sip.so to /etc/asterisk/modules.conf , same |
| 07:12.02 | gusto | i do not care about portage |
| 07:12.14 | fling | gusto: sorry |
| 07:12.22 | gusto | well, maybe he is ignoring your /etc/asterisk/modules.conf |
| 07:13.34 | fling | gusto: how to check this? |
| 07:13.36 | gusto | did you check out the /etc/init.d/asterisk? |
| 07:14.42 | fling | http://dpaste.com/766278/ |
| 07:17.58 | fling | gusto: i have only these config files > http://dpaste.com/766279/ ; sip.conf and extensions.conf are from http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client ; other are sample config files |
| 07:18.00 | gusto | that init.d script does not look good |
| 07:18.05 | gusto | it is too complicated |
| 07:19.37 | gusto | when your asterisk daemon is running, can you connect to it with 'asterisk -r' ? |
| 07:19.59 | gusto | or are you dropped to the asterisk console right after starting that daemon? |
| 07:20.31 | fling | I can connect |
| 07:20.40 | fling | and load the module |
| 07:21.00 | gusto | hmm |
| 07:21.31 | gusto | did you try it with onlu load = chan_sip or autoload=yes ? |
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| 07:22.18 | gusto | however, it seems to be a configuration problem, the only problem is to find where it is |
| 07:23.27 | gusto | that's why i do not use gentoo for 10 years now, i did use it when it begun, at the time gentoo was new, but later it became crap and apparently it still is |
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| 07:26.12 | oej_ | Start asterisk with "asterisk -cvvvv" and you will see all messages as chan_sip loads and fails. |
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| 07:32.15 | fling | < gusto> did you try it with onlu load = chan_sip or autoload=yes ? |
| 07:32.22 | fling | gusto: what should I try |
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| 07:52.07 | fling | oej_: it loads if I start asterisk directly |
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| 08:55.43 | fling | gusto: idk what to do next |
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| 09:05.29 | linocisco | who have done asterisk installed on wifi router? some said we can install it on Linksys WRT54G. But how about higher model or other brands? |
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| 09:12.13 | philfine | Hello everyone |
| 09:13.35 | philfine | DISA is giving me busy tone after I press the first number, even when the selected context has plenty on extensions |
| 09:15.06 | philfine | Example dial plan: http://pastebin.com/aadCqWCX |
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| 09:20.09 | hrolf | Can we do like initiate a call through AMI and then transfer it to FastAGI for further handling? |
| 09:25.56 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 09:26.07 | elliot98 | enters with a big wave |
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| 09:33.09 | pithagorians | hello. how is it possible to push the SDP parameters from asterisk to devices |
| 09:35.05 | pithagorians | i have an account that connects to asterisk by using TLS and SRTP. it comes registered, i can call from it but when i call TO it i get " == Everyone is busy/congested at this time (1:0/1/0) |
| 09:35.05 | pithagorians | " |
| 09:35.27 | pithagorians | it is an android phone and csipsimple as sip client |
| 09:36.05 | pithagorians | the devs of the project supposes that pushing SDP parameters from asterisk can help |
| 09:36.08 | pithagorians | any clye? |
| 09:36.10 | pithagorians | clue* |
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| 09:43.25 | sparf | Hi all! How to tell which dahdi-tools svn trunk revision is which official release version? |
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| 10:06.12 | elliot98 | when asterisk binds to multiple IP addresses, which source IP address does it take when responding to a request? |
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| 10:19.55 | catphish | does the CDR always hide the CLI of the caller where its set to withheld? |
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| 10:57.43 | hrolf | Can we do like initiate a call through AMI and then transfer it to FastAGI for further handling? |
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| 11:15.38 | hrolf | Can we do like initiate a call through AMI and then transfer it to FastAGI for further handling? |
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| 11:53.44 | catphish | am i correct in thinking that the mysql cdr loads a list of columns from the database when it loads and only inserts those columns that are present in the database? |
| 11:59.09 | WIMPy | elliot98: That depends on your routing table. |
| 11:59.22 | WIMPy | hrolf: Sure |
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| 12:21.47 | jrippon | Hi all, I am trying to install Asterisk1.6 on a up-to-date CentOS5 machine from the yum repositories provided by digium - it is failing because they depend on an older kernel version that the one installed, is this something to be expected and should I force my machine to downgrade its kernel or have I done something silly? |
| 12:22.12 | atan | There was this bluetooth device that would plug in to your network and allow you to use a mobile phone like an ATA adapter... anyone recall the name of this product? |
| 12:23.35 | WIMPy | jrippon: What about using a half way current version of Asterisk? |
| 12:23.59 | WIMPy | atan: What about chan_mobile? |
| 12:24.31 | atan | WIMPy, chan_mobile is interesting but I'm really wanting to know more about this little bluetooth device... |
| 12:24.57 | jrippon | WIMPy: I was just following the notes from http://www.asterisk.org/downloads/yum which told me to run "yum install asterisk16" - I'd prefer to stay within the package management if I can, is this out of date? |
| 12:25.03 | atan | It was talked about once in here and then I figured I would have bookmarked it, but no |
| 12:25.16 | WIMPy | atan: I haven't seen that as a device. |
| 12:26.09 | WIMPy | jrippon: Yes. Current versions are 1.8 (LTS) or 10 with 11 coming up. |
| 12:27.08 | jrippon | WIMPy: ahh the asterisknow repos are much more up to date it would seem, I'll give them a shot - thanks :) |
| 12:28.33 | WIMPy | Oh, and BTW: The kernel version only matters if you've got hardware interfaces that need drivers. |
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| 12:36.11 | jrippon | WIMPy: looks like I was indeed reading old info - thanks for you help :) |
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| 13:01.28 | elliot98 | WIMPy: a system that has two NICs connected to the Internet, Asterisk would need to choose the correct NIC to respond to |
| 13:02.01 | WIMPy | Asterisk doesn;t support multihomed environments. |
| 13:02.11 | elliot98 | WIMPy: the routing table is dependent on the source IP address, there is no default gateway |
| 13:02.15 | elliot98 | WIMPy: ah, ok |
| 13:02.21 | WIMPy | Unless that has changed recently. |
| 13:03.30 | elliot98 | WIMPy: that's true even for TCP? |
| 13:07.09 | elliot98 | WIMPy: so binding to all addresses would only work if the routing table is set up properly |
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| 13:13.13 | WIMPy | I don't see how TCp could go wrong, But I'm not sure what happens to RTP. |
| 13:13.33 | WIMPy | Last time I tried it went pretty wrong. |
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| 13:13.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:14.04 | elliot98 | so TCP IAX |
| 13:14.25 | WIMPy | IAX dopesn't support TCP. |
| 13:16.26 | elliot98 | why would TCP work with mutliple gateways? |
| 13:17.04 | WIMPy | It's a connection so it has a fixed source address. |
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| 13:19.13 | elliot98 | so the OS keeps a stack of all TCP connections, whereas UDP, perhaps only to keep the external port open? |
| 13:19.42 | WIMPy | UDP is connectionless. |
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| 13:20.36 | DND | hi guys in cli, how can i make the cli show live logs of specific extension? |
| 13:20.56 | elliot98 | but the OS could still keep track of ports using UDP and set the source IP accordingly |
| 13:21.08 | WIMPy | DND: No such thing. |
| 13:21.29 | elliot98 | for example, NAT keeps a port open to a specific IP address |
| 13:21.34 | DND | WIMPy, i see because its hard to diagnose specific extension if im being flooded with other extensions |
| 13:21.49 | WIMPy | elliot98: You can have "connectio" tracking in the netfilter, but this is usually done at application level. |
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| 13:25.44 | elliot98 | and the routing would not override whatever has been chosen on the application leve |
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| 13:30.33 | jacc0 | hi all! |
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| 13:32.14 | sparf | Hi! Can't I run dahdi_cfg from udev rule? |
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| 13:43.48 | tzafrir_laptop | sparf, basically, yes. Do you have more than one device? |
| 13:44.08 | tzafrir_laptop | What version of dahdi? |
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| 13:52.55 | fulcan | I am getting 'SIP/2.0 404 Not here' but it's because my carrier isn't getting the phone number to deliver the message and that's because I am a little confused as to how to give it to them. They request the format send(To:destination[From:sender]) and asterisk request messagesend(To:carrier[From:sender]), see the conflict? How to I make it so asterisk 'knows automatically' to send all sms to a certain destination/url/domain so my outbound form |
| 13:52.58 | fulcan | http://pastie.org/4177758 |
| 13:53.06 | fulcan | Can a static destination be configured in sip.conf so that messageSend will ignore the To: section? |
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| 13:59.12 | mjordan | fulcan: I don't understand your question. Given your example where the To: parameter is being passed "sip:19175039892@sip.anveo.com:5010", what does your provider want it to be? |
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| 14:12.39 | Azerus | eh |
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| 14:14.02 | jacc0 | hi |
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| 14:40.43 | fulcan | mjordan My To: is the carrier in format userNumber@carrier. but note there is no final destination number int that To:. But this is the only way asterisk will send the sms to my carrier. but my carrier needs to know where to send the sms. To:finalDestination in order to send. When someone thunk the technology up they forgot an entire field! |
| 14:42.25 | fulcan | It's 2 (one two) hand offs to get to final destination. Carrier is hop 1 and requires addressing to get to them and finalDestination from carrier is hop #2 and has it's own addressing. |
| 14:42.42 | fulcan | hops to get to final destination |
| 14:44.40 | mjordan | fulcan: first of all, its a SIP MESSAGE request. Its not an sms. |
| 14:45.35 | fulcan | mjordan I am working with SIP/sms |
| 14:45.43 | mjordan | fulcan: I have no idea what that is.' |
| 14:46.00 | fulcan | mjordan sms over sip protocol. |
| 14:46.05 | mjordan | fulcan: the MessageSend application is going to send a MESSAGE request. |
| 14:47.34 | fulcan | mjordan yup and If I used it correctly like To:2125551212(final destination)[From:"Joe"8182221313(origination), asterisk would have no clue to send this meesage to my carrier for delivery. |
| 14:48.13 | mjordan | sure. You have to specify a valid URI to send the message to. That's no different then any other SIP request. |
| 14:48.44 | fulcan | mjordan where? every attempt I have made failed miserably. |
| 14:49.43 | mjordan | The MessageSend application is going to set the To: header based on what you provide it. Whether or not that resolves as anything is subject to the usual constraints for sending requests out from Asterisk. |
| 14:50.09 | fulcan | I tried concatenating them together, using a colon, every format attempt flunked. |
| 14:50.10 | mjordan | so that could be a URI to send to, or a peer name |
| 14:50.16 | mjordan | fulcan: why would it succeed? |
| 14:50.41 | fulcan | mjordan I need a sample, I have been playing 'guess games' all weekend. I need a sample of usage. |
| 14:53.17 | fulcan | mjordan in what format? |
| 14:53.29 | mjordan | I'm not sure what you mean. |
| 14:53.43 | fulcan | mjordan what is it supposed to look like? sample? |
| 14:53.55 | mjordan | I'm trying to write one up for you, be patient please :-) |
| 14:54.05 | fulcan | thank you sooooo much. |
| 14:55.47 | mjordan | I'm not sure its going to help you. It sounds like you want SIP to do something different other then send a request to a URI. |
| 14:56.11 | mjordan | http://pastebin.com/1RdCSWff |
| 14:58.51 | fulcan | on line 22 it would fail because SIP:user must be the final destination, not the user. Is the sms being sent to yourself? user? |
| 14:59.21 | fulcan | To: is 'where the sms is going'. |
| 14:59.44 | mjordan | fulcan: think of this the same way as you would make an INVITE request to a UA. |
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| 14:59.59 | mjordan | except that you're sending a MESSAGE request instead. |
| 15:00.32 | fulcan | mjordan so, where to I pass the final destination number? nobody has told this sms where to go. |
| 15:01.04 | fulcan | to user? to carrier? nooooo, it has a LOT farther to go. where is this information placed into the message? |
| 15:01.30 | file | that's up to how they have decided how to use MESSAGE... |
| 15:02.31 | fulcan | file sms 101 message starts here and ends there. where to we specify 'there'? and not the inbetween folks either. |
| 15:03.48 | fulcan | I need to send a meesage to 'Joe'. Joe has the number 2222222222. Where in your sample would 2222222222 go? |
| 15:03.58 | file | MESSAGE isn't for SMS, it's for SIP messages - how your provider has decided to use it to implement SMS is up to them |
| 15:04.04 | file | it could use the request URI |
| 15:04.06 | file | it could use the To header |
| 15:04.09 | file | it could use a custom header |
| 15:04.33 | fulcan | file If you could show me how it would mean the world to me. |
| 15:04.44 | file | I don't know how your provider has decided to do it |
| 15:05.10 | fulcan | http://www.anveo.com/api.asp?code=apihelp_sms_receive_sip |
| 15:05.43 | file | that doesn't show it completely |
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| 15:06.54 | file | but MessageSend(sip:<destination phone number>@<peer entry in sip.conf for provider>) comes to mind |
| 15:06.56 | file | otherwise I have no idea |
| 15:07.38 | fulcan | file thats what I thought to but it doesn't work. one second, I will post the error. |
| 15:07.57 | file | without knowing exactly what the provider wants... I can offer no other input |
| 15:08.29 | fulcan | file your above example helps me a lot. at least I know I am not crazy. |
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| 15:37.48 | esaym153 | is there a way in the asterisk console to set everything to debug? |
| 15:38.22 | file | fulcan is gone... too bad! that page they linked is for receiving SMS messages over SIP, not sending them |
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| 15:57.04 | ectospasm | esaym153: make sure console in logger.conf has "debug", and then in the CLI "core set debug <DEBUG_LEVEL>" e.g. "core set debug 10" |
| 15:57.44 | ectospasm | if you have to change logger.conf, be sure to run "logger reload" first |
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| 17:04.54 | nutxase | anyone got mISDN working with el6? |
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| 17:08.30 | WIMPy | What is el6? |
| 17:08.34 | nutxase | cant seem to get my b410p working with centos 6 unless i use dahdi which drops calls |
| 17:08.41 | nutxase | WIMPy centos 6 |
| 17:09.23 | WIMPy | Ah, we had that a few weeks ago. Use a unmangled kernel from kernel.org instead of the RedHat one. |
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| 21:26.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), 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 |
| 21:30.21 | navaismo | ~pastebin |
| 21:30.21 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:30.38 | navaismo | the hangover is gone! |
| 21:31.57 | navaismo | hate you cisco call manager |
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| 22:10.10 | fulcan | I am trying to use Set(MESSAGE(body)=${FILE(/home/SMSbody.txt,0,140)}) in an sms message but appareantly there is an illegal escape character given when it tries to send. How do I properly escape '/home/SMSbody.txt'? |
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| 22:18.58 | fulcan | (that character is 0x10 AKA Line Feed which usually added by pressing Enter key) |
| 22:19.54 | fulcan | yet adding \n at the end of the message does not produce expected results??? |
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