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| 00:38.31 | Apteryx | Hello! I'm testing inbound calls, and it seems my ITSP trunk is not setup right, because when I'm dialing (from a test phone) my number, I get the notice: NOTICE[28395]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:17607058888@208.65.240.165>' failed for '208.65.240.44:5060' (callid: 32B8D779@208.72.120.66~o~o) - No matching endpoint found |
| 00:39.51 | Apteryx | I'm a bit puzzled. That is the endpoint it's trying to reach? Seems like it's "208.65.240.44:5060"? |
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| 01:51.15 | Apteryx | I've fixed my problem, it was quite stupid (there was no endpoint due to me using TCP as transport, while Asterisk was only configured to use UDP). |
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| 02:47.43 | ChannelZ | points and laughs |
| 02:49.02 | ChannelZ | (I kid, I kid) |
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| 03:41.14 | al_nz1 | afternoon peoples |
| 03:41.34 | al_nz1 | HI [TK]D-Fender |
| 03:42.12 | al_nz1 | Heya [TK]D-Fender - re my registration issue for a remote server - does my last post at the bottom of this thread look like a correct analysis |
| 03:42.12 | al_nz1 | http://pbxinaflash.com/community/index.php?threads/cisco-7941-piaf-on-ec2.15623/ |
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| 15:45.33 | darkdrgn2k | is there ANY way to have asterisk not intercept inband DTFM tones and regenerate them withouth butchering dsc.c |
| 15:46.14 | [TK]D-Fender | Allow them to reinvite |
| 15:47.07 | darkdrgn2k | its not working :( i set mediadirect to yes even mediadirectrtp to yes both are still proxying |
| 15:47.34 | [TK]D-Fender | Show comfings & debug |
| 15:47.40 | [TK]D-Fender | configs* |
| 15:48.16 | darkdrgn2k | i got it somewhat wroking be lowering the mindtfmtone ... |
| 15:48.40 | darkdrgn2k | mindtmfduration |
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| 17:47.32 | koffel | hello i am using asterisk 11.2 is there a way to cache register phone number into peers |
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| 18:05.00 | [TK]D-Fender | As in? |
| 18:06.27 | koffel | i using realtim |
| 18:06.29 | koffel | time |
| 18:06.38 | koffel | peers show up no problem |
| 18:07.07 | koffel | but i have to call out for sip phone number to be active |
| 18:11.01 | koffel | http://pastebin.com/hqTzgdsU |
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| 20:46.02 | skoenman | hey guys |
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| 20:48.49 | al_nz1 | Shouldnt my asterisk box be listening on 5060? |
| 20:48.56 | al_nz1 | netstat shows nothing for 5060 |
| 20:53.13 | skoenman | isnt that only when phones are registered?? |
| 20:53.34 | carrar | netstat -l | grep 5060 |
| 20:53.38 | skoenman | check sometimes they use different ports |
| 20:53.39 | ChannelZ | it's UDP mind you |
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| 20:54.37 | al_nz1 | ChannelZ: yeah, well I am trying to make my phone use TCP |
| 20:58.22 | al_nz1 | I have followed instructions |
| 20:58.35 | al_nz1 | but all i see on wireshark is UDP packets :-( |
| 20:59.28 | carrar | You need to configure Asterisk to listen on TCP 5060 first |
| 20:59.29 | carrar | tcpenable=yes |
| 20:59.30 | carrar | transport=tcp |
| 21:01.29 | carrar | tcpbindaddr=0.0.0.0 |
| 21:02.48 | al_nz1 | ok, i had added in asterisk sip tcpenable=yes, and tcpbindaddr=0.0.0.0 but not the transport=tcp line |
| 21:03.13 | al_nz1 | the tutorials I saw said to add the transport=tcp line to sip_custom_post.conf |
| 21:03.48 | al_nz1 | and to enable it for the extension in the gui |
| 21:03.58 | carrar | there is no sip_custom_post.conf |
| 21:04.24 | al_nz1 | ok, I will try in the asterisk sip menu to add it there |
| 21:04.46 | al_nz1 | how do I make it listen on 5060? |
| 21:04.53 | carrar | tcpbindaddr=0.0.0.0:5060 |
| 21:05.10 | carrar | 5060 is the default however |
| 21:05.12 | ChannelZ | carrar: in freepbx/etc there are all kinds of configs. anything you want it to be! :) |
| 21:05.27 | carrar | ChannelZ: Then he is in the wrong channel :) |
| 21:05.45 | carrar | #freepbx is what you want |
| 21:06.45 | ovoshlook | Hello. I have ISuue with Mix_Monitor. I need to write record_files only if call is answered. Now I Set Mix_Monitor(filename,d,option) but it still writes files with unanswered calls |
| 21:07.25 | al_nz1 | carrar: i will try those answers |
| 21:07.45 | carrar | You might download Asterisk and use that |
| 21:08.05 | carrar | there are example config files that have all that in them for you |
| 21:09.08 | carrar | http://www.asterisk.org/downloads |
| 21:09.53 | al_nz1 | carrar: fwiw I am trying to use TCP because I hope it will work around this problem: http://pbxinaflash.com/community/index.php?threads/cisco-7941-piaf-on-ec2.15623/ |
| 21:10.02 | al_nz1 | carrar: it seems to be a rport problem |
| 21:10.09 | ovoshlook | #asterisk-dev |
| 21:10.10 | al_nz1 | since my phone I dont think supports it |
| 21:10.47 | al_nz1 | the thoery was if I switch to TCP, I wont get the rport problem and thus it will work |
| 21:11.36 | carrar | You might use a better phone |
| 21:11.57 | al_nz1 | carrar: yeah, the ciscos are a pain in the ass |
| 21:12.04 | al_nz1 | but I have one now, and they are so common |
| 21:12.06 | carrar | I personally have a 7941 on my desk here |
| 21:12.18 | carrar | but they suck config wise |
| 21:12.27 | al_nz1 | carrar: i hear ya! |
| 21:12.56 | al_nz1 | I think my summary of the packet trace at both ends really shows the problem is likely to be rport |
| 21:12.59 | carrar | new polycoms are nice |
| 21:13.07 | al_nz1 | and TCP should fix |
| 21:13.21 | al_nz1 | but I have just tried your suggestions and still no joy |
| 21:13.29 | al_nz1 | maybe I need to reboot the asterisk server |
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