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| 01:34.32 | igcewieling | A sample pjsip.conf with sample Adtran config and sample Polycom config. https://pastebin.com/Aye6CmSa |
| 01:35.18 | velix | Hmm... Does anyone send OPTIONS to me?! http://dpaste.com/007YKPJ |
| 01:35.22 | igcewieling | Taken from my production configs, not identical but should be a usable config. |
| 01:35.55 | igcewieling | velix: it isn't normal for options to have a valid number, but other than that, it looks fine. |
| 01:36.16 | velix | igcewieling: Okay. Interesting. |
| 01:36.31 | igcewieling | not found is what you should expect. |
| 01:37.04 | velix | igcewieling: Really? I'm getting "200 OK" from some servers :D |
| 01:37.22 | igcewieling | velix: yup. really ANY reply should work. |
| 01:37.43 | velix | igcewieling: Ah, after removing `contact_user = 90754108935` I'm getting 200 OK ,too. |
| 01:37.52 | velix | typo in user sorry |
| 01:38.31 | igcewieling | It doesn't matter. The fact you are getting a reply proves the far end is alive and replying to packets. |
| 01:39.33 | velix | Ok. 404 just make me shrug. |
| 01:40.01 | velix | I've ported everything to pjsip now and it works fine. |
| 01:40.32 | velix | More than 80% of the calls through the University PBX worked today. |
| 01:41.08 | igcewieling | only 20% to go. |
| 01:41.50 | velix | igcewieling: I think the 20% are quick hangups on the other side (like 30-40 seconds after beginning the talk). |
| 01:42.02 | velix | igcewieling: I can't filter them out of the call log of course. |
| 01:42.25 | igcewieling | Do you don't actually know if the calls worked, you are just assuming based on call length. |
| 01:42.49 | velix | igcewieling: The other calls worked, since the interviews were more than 10 minutes and marked as "interview completed". |
| 01:43.15 | velix | I didn't get information on the last 20% by now :D |
| 01:43.24 | velix | But pjsip seems to do a good job. |
| 01:43.30 | velix | chan_sip did get amok on 403. |
| 01:43.37 | velix | pjsip just retries. |
| 01:45.12 | velix | Also `module reload res_pjsip.so` is very helpful. |
| 01:45.21 | velix | When having a bad config, it tells me what's wrong ;) |
| 01:45.52 | velix | But the chan_sip-2-pjsip converter from contrib is veeery buggy. |
| 01:45.58 | velix | It ofter forgets "sip:" |
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| 01:47.32 | velix | I wonder, why I didn't need "line=yes" on my incoming trunk. |
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| 02:04.12 | velix | Hmm, why does this happen? `Registration attempt from endpoint 'student_01' to AOR 'student_01' will exceed max contacts of 1` ? |
| 02:05.14 | igcewieling | chances are the far end NAT router closed the NAT translation, so when the phone re-registers it comes from a different port and creates a new contact. |
| 02:07.17 | velix | igcewieling: but the default value of contants is "0". |
| 02:07.20 | velix | contacts* |
| 02:11.32 | igcewieling | right, which means "unlimited" |
| 02:11.47 | igcewieling | Why did you set it to 1? |
| 02:13.50 | velix | igcewieling: oops, seems like I've missread it. Thought default is ZERO. |
| 02:13.57 | velix | don't say a word. |
| 02:13.59 | velix | please :) |
| 02:15.28 | velix | OT: I've just seen "Star Trek Insurrection" again. "Software Support: PIXAR Animation" :D |
| 02:16.29 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-aor_max_contacts |
| 02:17.16 | velix | igcewieling: Ah, I remember what I did. I've left type=aor blank and got an error. |
| 02:17.21 | velix | Then I've set it to "1". |
| 02:17.37 | velix | When setting it to zero, I can't register my client |
| 02:18.34 | igcewieling | set it to 1 and add remove_existing set to yes and move on with life. |
| 02:19.51 | velix | That's something we'd need to at to sip_to_pjsip :D |
| 02:19.56 | velix | to add* |
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| 06:49.46 | muks | folks, what happens if multiple clients authenticate to the same endpoint (using same username/password)? |
| 06:49.51 | muks | SIP clients |
| 06:52.00 | muks | can they function indepdently similar to jabber/XMPP "carbons" feature? what happens if the endpoint is dialled from extensions.conf.. are all the clients authenticated to that endpoint rung? |
| 06:53.10 | igcewieling | muks: that depends on what sip driver you are using. |
| 06:55.53 | drmessano | Never heard it called "carbons" |
| 07:05.59 | muks | pjsip |
| 07:06.57 | muks | drmessano: i guess it comes from carbon-copy |
| 07:07.01 | muks | https://xmpp.org/extensions/xep-0280.html |
| 07:07.25 | yang | Hello, I have a problem with incoming calls. I am trying to call +38617774515 which is a number allocated/forwarded to me by the SIP provider (localphone.com). The phone on my side doesn't ring. It can also be a firewall issue? I can mention that I am able to do outgoing calls through the same SIP account (provider). Here is the debug log http://paste.debian.net/hidden/345b13fe |
| 07:07.33 | yang | I can also see the call being received in "debug" From: "38669944499" <sip:38669944499@95.211.119.240>;tag=11-2D00F42F-5EA29DAF00012FE0-D1D9D700 - To: <sip:38617774515@localphone.com> |
| 07:07.49 | drmessano | muks: The actual multiple connections are resources, not carbons |
| 07:08.02 | drmessano | AORs in SIP |
| 07:08.41 | muks | drmessano: carbons broadcasts messages to all resources |
| 07:09.27 | drmessano | The multiple authenticated clients are "resources", akin to SIP AORs |
| 07:10.17 | muks | with the pjsip driver, what happens if the extension is called, when it has 2 connected clients? |
| 07:10.38 | drmessano | They all get rung unless you further specify an AOR |
| 07:11.59 | muks | drmessano: thank you for that answer.. i will read about AORs |
| 07:14.09 | drmessano | Well, if you use the standard PJSIP/exten dial it will ring the first AOR |
| 07:14.18 | drmessano | There's a modifier to dial all AORs |
| 07:14.33 | drmessano | Which is commonly used |
| 07:16.04 | muks | perhaps a reading of RFC 3621 is due |
| 07:16.27 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 07:16.31 | drmessano | Not really |
| 07:16.47 | igcewieling | In the aor section for the endpoint, add max_contacts=X where X is the max contacts you want to allow |
| 07:16.49 | drmessano | There's a whole wiki |
| 07:16.58 | muks | i've spent a lot of time on configuring asterisk this year, and it works fine as a company PBX, but i don't know anything about the protocol |
| 07:17.09 | muks | i.e., SIP |
| 07:17.25 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 07:19.08 | muks | drmessano: thank you.. i should use PJSIP_DIAL_CONTACTS() and remove per-device endpoints that i'd setup |
| 07:19.32 | drmessano | Depends on the need |
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| 08:15.01 | yang | My question is, how do I make my phone ring and answer the incoming call ? |
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