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| 03:29.00 | janicez | meowsterisk |
| 03:29.07 | janicez | [TK]D-Fender: How are you? |
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| 05:08.51 | ChannelZ | comatose |
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| 05:54.22 | janicez | ChannelZ: same |
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| 07:18.56 | Geek-Linux | Hi. I have moved my asterisk server behind firewall but now i am facing the muting issue in SIP calling. I cant receive RTP packets of user on my server. thank you in advance. |
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| 08:38.18 | janicez | Geek-Linux: check your firewall settings |
| 08:40.04 | ChannelZ | You need to configure externaddr and localnet in sip.conf; you need to port-forward the range of RTP ports as configured in rtp.conf to the asterisk box |
| 08:40.35 | ChannelZ | And peers outside the firewall who themselves are behind firewalls need nat turned on for them |
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| 10:07.16 | nilesh | hello |
| 10:07.21 | nilesh | anybody using cisco 7940 here? |
| 10:18.10 | MaliutaLap | yes |
| 10:25.51 | nilesh | MaliutaLap: do you use chan_mobile? |
| 10:26.02 | MaliutaLap | no |
| 10:26.03 | Geek-Linux | <janicez>: Firewall is currently natted open for every connection to the server. |
| 10:26.24 | janicez | Geek-Linux: My nick doesn't have <> around it. |
| 10:26.24 | MaliutaLap | nilesh: why would chan_mobile have anything to do with a cisco 7940? |
| 10:26.32 | janicez | Geek-Linux: just type jan and then the tab key |
| 10:27.14 | Geek-Linux | ChannelZ: externaddr is the public ip that has been natted to the server ? |
| 10:27.17 | MaliutaLap | nilesh: usually the 7940 is used with chan_sip (/pjsip) or chan_skinny |
| 10:27.18 | nilesh | MaliutaLap: well nothing... this chan_mobile issue has been a PITA for me. I'm unable to hear whatever the other side says on a hardphone, but the same thing works with a softphone. |
| 10:27.36 | nilesh | MaliutaLap: given the codec is same on hardphone and softphone (ulaw/alaw) |
| 10:28.17 | janicez | nilesh: uh |
| 10:28.31 | nilesh | janicez: where could be the issue?? |
| 10:28.39 | janicez | nilesh: not a clue in hell |
| 10:28.46 | janicez | nilesh: probably a phone side issue |
| 10:28.59 | janicez | nilesh: try calling another extension |
| 10:29.08 | nilesh | janicez: but how two different hardphones can have the SAME issue?! |
| 10:29.17 | janicez | nilesh: bug in their code? |
| 10:29.32 | nilesh | janicez: different brands man, Cortelco and Cisco |
| 10:29.42 | nilesh | inter-extension calling works fine |
| 10:29.47 | janicez | nilesh: they may still be the same hardware; do they look nearly the same? |
| 10:29.53 | janicez | nilesh: What version of asterisk are you using? |
| 10:29.57 | nilesh | of course not, asterisk 13 |
| 10:30.20 | janicez | nilesh: no clue then |
| 10:31.16 | robmal | Oh, a problem. |
| 10:31.21 | robmal | What's going on? |
| 10:32.03 | nilesh | robmal: no audio on hardphone when used with chan_mobile, but works on softphone. codecs are same (ulaw/alaw) |
| 10:33.04 | robmal | Show me the pcaps |
| 10:33.25 | nilesh | I can just hear some noise on the hardphone. |
| 10:33.51 | nilesh | whenever the other side says something |
| 10:33.53 | robmal | Also, which way are the calls made? From the pbx to mobile or from mobile to pbx? |
| 10:34.02 | nilesh | both way same behaviour |
| 10:34.17 | robmal | Impossiburu. |
| 10:34.25 | robmal | Again, show me the packet captures. |
| 10:34.52 | nilesh | yes wait, generating |
| 10:45.25 | nilesh | robmal: https://nileshgr.com/asterisk.pcap |
| 10:45.45 | nilesh | in the dump, 172.16.0.8 is ekiga running on windows machine, and 172.16.0.5 is cisco 7940 |
| 10:46.03 | nilesh | everything is clear when I'm using ekiga |
| 10:46.07 | MaliutaLap | So why is chan_mobile even being used |
| 10:46.19 | MaliutaLap | they should both be on chan_sip |
| 10:47.02 | nilesh | MaliutaLap: because I want outgoing calls to happen via mobile phone? |
| 10:47.16 | nilesh | outgoing / incoming, basically trying to set up PBX using a mobile phone |
| 10:53.33 | robmal | Ok, anything happening in dmesg? |
| 10:53.45 | robmal | And if you have, paste your asterisk log. |
| 10:56.10 | nilesh | robmal: Bluetooth: hci0 SCO packet for unknown connection handle 1 | lots of such messages |
| 10:56.13 | nilesh | in dmesg |
| 10:58.27 | robmal | Do you have any other usb dongles you could check? |
| 11:00.12 | nilesh | nothing interesting in asterisk logs : https://gist.github.com/anonymous/87ef13482fe00148f5bf |
| 11:00.26 | nilesh | robmal: I have two usb dongles of the same model, I even tried alignmentdetection=yes, but doesn't help |
| 11:00.38 | nilesh | the only thing left now is to try this on a laptop |
| 11:01.49 | robmal | Ye, looks like some kind of driver issue, try it on a laptop. |
| 11:02.44 | nilesh | I don't think the BT dongle can be an issue because the things get recorded fine! and also if it was a dongle issue how does a softphone work? |
| 11:02.58 | nilesh | so that rules out the dongle issue |
| 11:03.10 | nilesh | this is something to do with codecs |
| 11:03.22 | nilesh | I don't know how to debug this :( |
| 11:04.10 | nilesh | robmal: you found anything interesting from the pcap? |
| 11:05.37 | robmal | Nope, both invites are sending the same codecs. |
| 11:07.10 | nilesh | this is frggin ridiculous. Same codec, same bt dongle, same mobile. But works with softphone and doesn't work with hardphone.!??!? |
| 11:07.53 | MaliutaLap | nilesh: is there anything different with the SIP config for the cisco vs ekiga? |
| 11:07.59 | nilesh | MaliutaLap: nope |
| 11:08.15 | nilesh | In fact I had to disable authentication and use identify by ip for cisco |
| 11:08.27 | nilesh | otherwise it wasn't registering |
| 11:08.44 | robmal | Did you try using ext 11 on cisco? |
| 11:09.07 | nilesh | robmal: what do you mean? |
| 11:09.45 | nilesh | if you are just pointing to a different extension, then it won't make any difference because the configuration is a template for endpoints |
| 11:10.24 | robmal | Meh. |
| 11:10.42 | robmal | Any other hardphones available? |
| 11:11.09 | nilesh | no, only these two. In fact I ordered cisco because I thought the cortelco ones I had were bad. |
| 11:15.02 | nilesh | ? |
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| 11:24.25 | nilesh | let me give a shot on the mailing list |
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| 15:06.52 | robmal | Hmm, any of you tried running an old rotary phone with a voip gateway? |
| 15:13.08 | [TK]D-Fender | I've known others who have |
| 15:14.27 | robmal | Did they make their own pulse to dtmf converters or did they buy them? |
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| 15:37.21 | [TK]D-Fender | The ATA they used did it for them |
| 15:37.28 | [TK]D-Fender | Don't recall for sure which. |
| 15:40.03 | robmal | I hope it was PAP2t |
| 15:40.04 | coppice | why would a converter be needed? |
| 15:41.06 | robmal | Rotary phones sent the number by pulses (pauses in the voltage) not by tones (changes in amplitude) |
| 15:43.24 | coppice | yes, we know, but why would a converter be needed? |
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| 15:53.51 | robmal | Because most gateways don't understand pulse and those who do not always work properly afaik. |
| 15:54.39 | coppice | I haven't met a gateway that won't work with pulse dialing, although there are probably some where they just don't test it well enough these days |
| 15:55.24 | coppice | its just so easy to do that it makes no sense to leave it out |
| 15:56.01 | coppice | not everything bothers to provide pulse outdialing these days |
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| 16:21.53 | krenel | hi |
| 16:26.02 | krenel | we just started using dpma and loaded a custom logo, but now we get "Digium Asterisk" at the top... is there a way to change that to the users actual name? |
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| 17:46.40 | krenel | nobody talks in here? |
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| 17:56.47 | gusto | I have a good question |
| 17:57.07 | gusto | how can I tell asterisk to register with the port that he is connecting with? |
| 17:57.29 | gusto | or maybe rport on the other-side's asterisk? |
| 17:58.44 | [TK]D-Fender | gusto, Your logic is backwards |
| 17:58.50 | gusto | yes |
| 17:58.51 | gusto | cool |
| 17:58.52 | gusto | :-D |
| 17:58.52 | [TK]D-Fender | first, what is "he"? |
| 17:59.07 | gusto | I have two asterisks over TLS connected with each other |
| 17:59.11 | [TK]D-Fender | And "hes" is connecting... so why is ASTERISK registering? |
| 17:59.19 | gusto | w8 |
| 17:59.31 | gusto | one is server and one is client, so far so good |
| 17:59.43 | [TK]D-Fender | Also, "connect" is too vague a term |
| 17:59.50 | gusto | but, the client is connected over a high port to a 5061 port of the server |
| 17:59.52 | [TK]D-Fender | Both are clients |
| 17:59.54 | [TK]D-Fender | ~b2bua |
| 17:59.54 | infobot | well, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 17:59.55 | [TK]D-Fender | ^^^ |
| 18:00.11 | gusto | yes, that's the problem that they are both clients and both servers |
| 18:00.35 | gusto | but with a cisco ata or a snom telephone it is not the case |
| 18:00.59 | gusto | for example my cisco ata is connecting with some high port and accepts also replys on that port |
| 18:01.55 | gusto | with the asterisks it is not the case, it is always from a high port to a low port (5061) no matter who is what |
| 18:03.15 | gusto | so there are two connections for "one" connection |
| 18:04.40 | gusto | http://pastebin.com/NFy3A9uz here |
| 18:04.41 | [TK]D-Fender | No, ALL are effectively "clients" |
| 18:04.47 | gusto | well |
| 18:04.49 | gusto | whatever |
| 18:04.57 | gusto | take a look at that paste |
| 18:04.58 | [TK]D-Fender | ATA's can have MULTIPLE ports because they are MULTI-PORT to begin with |
| 18:05.06 | [TK]D-Fender | Like those wth 2 ports. |
| 18:05.16 | [TK]D-Fender | So Each has to be addressed SEPARATELY |
| 18:05.52 | gusto | yes |
| 18:05.54 | gusto | of course |
| 18:06.26 | gusto | ata's working perfectly |
| 18:06.37 | [TK]D-Fender | And you're still talking about other things that don't matter |
| 18:08.03 | gusto | well, I understand that asterisks may get more connections over time, but I only need one connection from one asterisk to another asterisk, so is there a way how to get that asterisk to send the port he is running with? |
| 18:08.15 | gusto | running that concrete connection with |
| 18:08.44 | gusto | or maybe is it the case that the atas and phones do not send any port? that can also be the case... i have to check that out |
| 18:11.33 | krenel | Ping? |
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| 18:16.14 | [TK]D-Fender | krenel, relatively few people have Digium phones. Even fewer use DPMA with them. Fewer still are going to be on during the weekend |
| 18:16.50 | krenel | NP I just wanted to make sure my messages were actually going through! :) |
| 18:17.02 | krenel | So DPMA isn't widely used, you'd say? |
| 18:17.10 | [TK]D-Fender | krenel, You will probably have to check in once more people likely to have your answer are around. Like around Digium's working hours would be best |
| 18:17.51 | [TK]D-Fender | gusto, You are failing to understand that TLS being TCP has a SOURCE PORT and a DESTINATION port. |
| 18:18.08 | [TK]D-Fender | When the sender starts the connection, since each one is unique... that it will have a different SOURCE port |
| 18:18.19 | [TK]D-Fender | You CAN"T make them all the same otehrwise a system couldn't support multiple connections |
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| 18:20.03 | gusto | well, rport seems to do the trick |
| 18:20.18 | gusto | i have to test it as well |
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| 18:22.26 | gusto | hm, it works |
| 18:23.04 | gusto | [TK]D-Fender, there was never multiple connections |
| 18:23.47 | gusto | here in this paste http://pastebin.com/NFy3A9uz the first two connections were actually one connection only one was sending and the other one for recieving and i wanted to have that both in one port |
| 18:23.49 | gusto | eh |
| 18:23.56 | gusto | in one ESTABLISHED: ESTABLISHED |
| 18:24.06 | gusto | in one TLS connection not split into two |
| 18:26.12 | gusto | *** this is cool, it works |
| 18:26.39 | gusto | complex problem simple solution |
| 18:27.16 | gusto | and I am all day watching traces trying to figure out wtf is he doing |
| 18:28.09 | gusto | i already tried that rport, but i forgot to make sure that registrations disappear, so that was a false negative |
| 18:28.43 | gusto | now i kicked the peers out and as soon as they did re-register they had the "right" port on |
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