I am getting following warning very frequently on Asterisk 18
[Aug 17 15:01:26] WARNING[8722][C-00000337]: chan_sip.c:8153 sip_indicate: Don’t know how to indicate condition 36
That would be stream support related, which is not supported in chan_sip and can be ignored.
What to do to resolve this ?
You can ignore it. If there’s actually a problem, then you should state what the problem is. A warning message that can be ignored isn’t a problem itself.
SIP Trunks are getting Unreachable
That warning message is completely unrelated to SIP trunks going unreachable. Going unreachable occurs when chan_sip sends a SIP OPTIONS and gets no response. You would need to examine the network traffic, such as through a packet capture or “sip set debug on” and see if a SIP response is occurring.
As well just a note that chan_sip has been removed from Asterisk 21, receives no bug fixes, and no issues can be reported against it.
I am Dialing One server from Other using SIP, should I use some other mechanism like IAX2
SIP or IAX2 works for connecting Asterisk instances.
What should I do, Very frequently all servers gets unreachable, then We need to restart Asterisk to proceed, It is a Major Issue, Please help
I’ve told you what has to be done:
“You would need to examine the network traffic, such as through a packet capture or “sip set debug on” and see if a SIP response is occurring.”
It’s also possible that chan_sip itself has a problem, and as I stated that would not get looked into.
Should I swich to IAX2 instead of SIP
It may or may not change things. You can try it, it’s up to you.
What is the best practice to DIal one server to Other.
We’re going in circles at this point. SIP or IAX2 can be used. Many people use SIP, but ultimately it’s up to each person what they choose. The chan_sip module is no longer supported. The current SIP channel driver is chan_pjsip.
Ok So should we dial Other Servers Using pjsip
Try changing your port from 5060 to a different port. I too had this issue and was able to resolve changing port 5060
You can use the currently supported implementation of SIP, which is chan_pjsip. However, your problem seems to be network related rather than protocol or channel driver related.
if did not get it chan_sip has been replaced by chan_pjsip
so if you want to continue useing “RFC 3261” you should change to chan_pjsip
(sorry but link is broken, belive they are still migrating from old wiki)
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
The migration already happened. The link is here, Migrating from chan_sip to res_pjsip - Asterisk Documentation and if formatting issues are encountered then it is best to report them on Github at Issues · asterisk/documentation · GitHub
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