When someone calls in there is no ring back. The extension dialed will ring, but there is no indication of that for the person who called in. I'm not sure where to begin with this one so any ideas at all will be helpful. Thanks Jason
On the Inbound route you might need to set Signal RINGING before answer, especially if going through an IVR/Queue.
But if you answer, the audio in the call is fine, or you can't hear audio at all? Maybe it is a rtp problem.
It's very possible. Sometime, the operators have need this information (If I remember well, it's information 180 to SIP protocol, no?). But, I don't remember, in the case where you have an announcement into your "Ring Group", if you lose the announcement, you can have only the ring back tone!
Hey ya Danardf: I believe that if the phone sends a 183 (early media available) instead of a 180 then it is responsible for sending "ring-back" in rtp packets, if it sends a 180 then the calling party should supply it. signal ringing prompts asterisk to send a 180 if the endpoint(s) is "mis-behaving" after sending/receiving a 183. This part of Sip is often used to provide locale specific "ring-back" etc. Who knows in this situation?
I was not realy sure with the 180 Dicko. :huh: They should make some test and compare the dumps SIP. But i think that the info 180 or 183 from one side like the other, are involved, no? Are you ok with me?
I agree Franck, only a good debug will reveal, I definitely think that there is a 183 not being honored somewhere in the mix. my guess is either the phone or the provider, not asterisk.
ramoncio I do not seem to be having a problem with sound. Both parties can hear each other just fine. danaedf and dicko What do I need to do to check to see if what I need is getting to the right place? I think both of you agreed that some sort of debug needs to be done, but do I need to run debug on Elastix or do I need to do some packet capturing? Thanks Jason
If my original suggestion didn't work and exploring the phone setup doesn't give any insight then sip debug peer XXX from the asterisk CLI will show the SIP call progression, they go fast so sometimes it's better to explore them in var/log/asterisk/full "post mortem". you should be looking for lines that have "SIP/2.0 183 . . " and "SIP/2.0 180 . . ."
I do get a 180 message but I am not seeing a 183. Here is my 180 message: <--- SIP read from X.X.X.X:X ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X:X;branch=z9hG4bK6ebcc707;rport From: "Anonymous" <sip:NXXNXXXXXX@X.X.X.X>;tag=as52718d4e To: <sip:521@X.X.X.X>;tag=E8393ECB-5B4DE66C CSeq: 102 INVITE Call-ID: 06af6c6e4f0505d813d5459b7f7aa5a1@X.X.X.X Contact: <sip:521@X.X.X.X> User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 Allow-Events: talk,hold,conference Content-Length: 0
When you get that message that says the phone is ringing but it is not, then you will need to get the polycom manual out and start reading . . . I use the cheaper ones and have never had a problem with them, (I always default them before use)
Sorry for the delay. This phone system is not a high priority right now so I'm only able to work with it when I can find the time. The problem is not that the called phone doesn't ring, it does, the problem is that the person calling in does not have a ring back sound or any sound for that matter. When the phone is answered then it works fine.
I understand that the instrument is ringing, I should have said in my last post .. when you see the message that the phone is ringing, but there is no ringback. . . you might need to go to the manual. As I suggested, try resetting to factory default. To eliminate any VOIP provider problem, send the incoming to a ring group instead and check behavior
Well I got it working. I had to add progressinband=yes into the peer details. Thanks for your helps Jason Also congrats dicko on being the community member of the month.
