180 Ringing Signal Not Forwarded to Softphone

[dpkbarala]

Hi Asterisk Community,

I’m having an issue with my Asterisk server not forwarding the 180 Ringing signal to my softphone. The server receives the 180 Ringing signal but does not send it to the softphone. I’ve tried various combinations of prematuremedia and progressinband settings in sip_general_custom.conf without success.
I’ve checked network connectivity, enabled SIP debugging, and reviewed logs but still can’t resolve the issue. Attached is a sngrep screenshot showing the 180 Ringing signal being received. Any help or suggestions would be greatly appreciated!

[david551]

@dpkbarala

sip_general_custom.conf

That file is not referenced by Asterisk unless it is included from somewhere else. I suggest you ask this on the forum for the software that provides that include.

You haven’t provided any logging, and you haven’t provided the dialplan. It appears that the dialplan has done something that is causing call progress to be sent as early media. If you didn’t write the dialplan, you should consult whoever did.

Note that Asterisk doesn’t forward 180 ringing. It generates an internal, AST_CONTROL_RINGING message, which is then converted into whatever is needed on the calling channel. If both are using SIP and the caller has neither been answered, nor put into early media mode, that may look like forwarding, but it is actually a completely new request.

[dpkbarala]

thank you david for reply
i understand your reply and remove third party gui but now facing also same problem
operator sending us 183 and 180 but my server only sending 183 to caller.
thanks .
[mydialplan]
exten => _X.,1,Set(CALLERID(all)=XXXXXXX)
same => n,Noop(this is a test call ${EXTEN})
same => n,Dial(SIP/Jiotrunk/0${EXTEN})
same => n,Hangup()

[jcolp]

Pretty sure that’s the way chan_sip works in that scenario.

[dpkbarala]

hello jcolp
i am calling from my softphone to another server as of now via(iax2-trunk) and second server has connected to operator via sip trunk and operator sending us 183 and 180 both but my second server only sending 183 to first. and i dont now what happen with 180 ring signal that we received from operator.

[jcolp]

The chan_sip module doesn’t support sending a 180 after sending a 183.

[dpkbarala]

thanks jcolp,
for instant reply is there any way that we capture both signal and send it to caller or that’s all about it.

[jcolp]

I don’t understand the question.

[dpkbarala]

sorry for that i mean is there any way that if operator sends us 183 and 180 both than we can send it as it is to our caller(softphone). or second only can send 180 for same scenario.

[jcolp]

chan_sip doesn’t support sending 183 and then 180 to the best of my knowledge, as I’ve stated before. Asking multiple times doesn’t change that.

As for sending 180 only, there is the ‘r’ option to Dial which may do that, otherwise any behavior things would be the options in sip.conf.

That’s the extent of my recollection of chan_sip.

[dpkbarala]

i use r and R both option in dial that will send ring signal even callee is switch off or unreachable so i think this is not a good approach.

[david551]

You have to realise that chan_sip is dead, and very few people remember details of how it works.

Also, I seem to remember that not going back to 180 after 183 is part of the RFC on interworking with ISDN. In practice, most PSTN calls will end up interworking with ISDN at the provider.

I actually thought that I’d read that you needed to explicitly call Progress, for 183 even to, in effect, pass though, so, I wonder, as well as using an ancient channel driver, you are also using an ancient Asterisk. In reality, none of the SIP signally passes through; it gets converted to similar internal messages, and converted back.

[david551]

I couldn’t find an RFC reference to back up any limitations on the order of 180 and 183, and, if anything, the restrictions are the other way, but it seems like this has been a controversial area, even for chan_pjsip, 180 Ringing after 183 Progress is not passed on to the caller references to changes two the behaviour, in this respect.

[jcolp]

Aye, the functionality is behind an explicit option in PJSIP that has to be enabled so as to preserve prior behavior.

[dpkbarala]

hello david
i have switch asterisk version to 18.9.0 and make changes in chan_pjsip.c after that still my problem not solved can you address me for this because i still receiving 183 and 180 but sending only 180 to softphone.
thanks a lot.

[jcolp]

The version 18.9.0 does not include support for this, as it’s an old release of 18. If you’re making code changes to an old version to try to do this, then you’re on your own for that.