Difference between revisions of "Call hangs after 30 seconds"
From Kolmisoft Wiki
Jump to navigationJump to search
| Line 45: | Line 45: | ||
---- | ---- | ||
See also: | See also: | ||
* [[Maximum retries exceeded on transmission]] | |||
* [http://docstore.mik.ua/univercd/cc/td/doc/product/voice/ata/ataadmn/sip30ad/sip88aph.htm#wp1006974 SIP call flow] | * [http://docstore.mik.ua/univercd/cc/td/doc/product/voice/ata/ataadmn/sip30ad/sip88aph.htm#wp1006974 SIP call flow] | ||
* [[SIP INVITE]] | * [[SIP INVITE]] | ||
[[Image:sip_call_flow.png]] | [[Image:sip_call_flow.png]] | ||
Revision as of 14:18, 13 August 2009
- Check timeout - maybe call can't last more then 30s?
- Make sure call is made over SIP protocol
- Write down all IPs of all devices which participate in the session
- Enable sip debug (only for parties in the call, not global sip debug)
- Look for BYE packet, who is sending it.
- If there is no BYE packet - big chance that problem is in the network - e.g. BYE packet does not reach our end or other end has problems with SIP protocol implementation
- Check IP in SIP debug and compare to real IPs
- If it is possible - try to capture SIP debug at the other end and compare to SIP debug in MOR
If during the call you see similar line:
[Aug 13 14:19:34] NOTICE[15712]: chan_sip.c:15683 sip_poke_noanswer: Peer '148' is now UNREACHABLE! Last qualify: 0
That means that device become unreachable and connection lost. It should point to network problems.
Such lines:
[Aug 6 11:50:43] NOTICE[22403]: chan_sip.c:2927 auto_congest: Auto-congesting SIP/prov253-0a1966d0 -- SIP/prov253-0a1966d0 is circuit-busy Scheduling destruction of SIP dialog '0ceffd014862f2c043209aa56d4930f4@82.194.82.123' in 32000 ms (Method: INVITE)
Clearly points that there is no response from dialed server, so timeout occurred.
Possible solutions:
- Check IP/port settings in configuration
- Check network settings (bad routing/bad settings/etc)
- Change device at the other end
See also:
