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03:49.07 | apb1963 | asterisk-13.10.0 fails to install on debian - jesse. I don't know what to do about it. Details here: http://pastebin.com/ZxrSjG9t and here (search for "tdy"):https://bbs.archlinux.org/viewtopic.php?id=156517 |
03:54.48 | apb1963 | Here's the complete output of ./configure if it's of interest... http://pastebin.com/rCw0bETt |
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04:27.02 | voipmonk | hrmm |
04:28.47 | voipmonk | do you have /usr/lib/pkgconfig ? |
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04:30.24 | apb1963 | voipmonk: checking.... |
04:30.36 | voipmonk | apt-file search ptlib-config |
04:31.07 | apb1963 | <PROTECTED> |
04:31.09 | apb1963 | total 92 |
04:31.16 | voipmonk | excellent |
04:31.20 | apb1963 | [snip] |
04:31.26 | voipmonk | export PTLIB_CONFIG=/usr/lib/pkgconfig |
04:31.28 | voipmonk | ./configure |
04:33.09 | apb1963 | voipmonk: we have a weiner. Thank you! |
04:33.16 | voipmonk | bows |
04:33.39 | apb1963 | I take it someone will make sure it gets into the next version? :) |
04:33.59 | voipmonk | whats next? |
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07:25.16 | jeffspeff | how can i verify that sip tls is working? my phones sip peer transport is only set for tls. the transport option in sip.conf is set for udp, tls. tcp is enabled and tls is enabled in sip.conf as well as the certfile, cafile, cifer and clientmethod. i can make calls to the server from my peer. i assume that sip tls is working, but how can i be 100% sure? |
07:26.35 | Samot | You run some debug or sip captures. |
07:26.58 | Samot | Make sure it is doing TLS and the encryption is there. |
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07:28.45 | jeffspeff | I just got this error in the console, yet doing "sip show peers" still shows my peer status as ok and a good response time. ERROR[7195]: tcptls.c:945 ast_tcptls_client_start: Unable to connect SIP socket to MY-PHONE-IP-ADDRESS58192: Connection timed out |
07:30.07 | Samot | Show a call. |
07:30.10 | Samot | ~pb |
07:30.13 | infobot | hmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
07:33.31 | jeffspeff | i wasn't able to reproduce that error on the 2nd call. but i do have debug logs. let me sanitize the data... |
07:34.26 | Samot | Not logs. Live data. |
07:34.30 | Samot | Logs don't catch it all. |
07:35.00 | jeffspeff | I ran "sip set debug on" from the console and copied all the output |
07:38.12 | Samot | ok |
07:40.05 | jeffspeff | here ya go http://pastebin.com/m9e7BQa4 |
07:50.37 | drmessano | If the peer is set for TLS it's only going to do TLS |
07:52.21 | jeffspeff | ok, but what would cause that connection time out error i posed above? |
07:53.05 | jeffspeff | *posted |
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07:55.14 | drmessano | Inability to connect to the device. |
07:55.22 | drmessano | Is the device on the same LAN? |
07:57.01 | jeffspeff | no |
07:57.28 | jeffspeff | the server is on google cloud, phone is a bria softphone on my laptop |
07:59.22 | drmessano | Okay, sounds like your NAT closed the connection. |
08:00.25 | jeffspeff | as in the google firewall or my server. i can fix the server (i have root access). i can't change the firewall. lol |
08:00.40 | jeffspeff | *the google firewall |
08:00.43 | drmessano | What do you expect to fix on a firewall? |
08:01.01 | drmessano | Your client timed out.. there is no checkbox on a firewall to tell it not to do that |
08:01.05 | jeffspeff | nat time outs. |
08:01.21 | drmessano | Thats not how it works |
08:01.49 | jeffspeff | i thought you were saying it was a nat timeout issue. as in the firewall closed the nat port. |
08:02.08 | drmessano | It did.. when your client timed out |
08:02.19 | jeffspeff | ok, got ya |
08:02.49 | jeffspeff | that would explain why i haven't seen it again since. |
08:03.19 | jeffspeff | i'm setting up sip tls for the first time and just trying to cover everything |
08:03.51 | jeffspeff | my ISP is pretty bad with latency. i can completely understand if my client closed first. |
08:04.31 | jeffspeff | thank you for clarifying that. |
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22:00.00 | c|oneman | does anyone know if I can use a SIP service that uses Digital digest authentication instead of a SIP username and password |
22:00.38 | c|oneman | the credentials (email/password) are somehow converted into a hash used for auth in the native proprietary sipp app that uses this |
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