OBI100 SIP URI call problem
DavidNewton01:
I use an OBI100 as a regular ATA to register two sip servers. I have been using SIP URI call directly to the OBI100 for some time from sip devices or from CallCentric DID without problem. But recently, the SIP URI call does not work anymore. In "Call Status" page, the SP1 state shows as "unknown" while Phone state shows as "Connected" after the phone is picked up with one-way audio (incoming ok, no outgoing audio), if the call is from Callcentric or other sip devices. But if the call is from an asterisk server, then the call is ok (Both SP1 and Phone states show as "connected" in the call status page). There is no problem for this OBI100 to have calls from its registered SIP servers. I have forwarded the sip port and the rtp port range to this ATA in the DD-wrt router. Anyone has idea about this problem? Thanks in advance.
DavidNewton01:
Since nobody replied this topic yet, I would like to make my question simpler: what could cause this situation: in the call status page in OBI100, while the Phone State shows "Connected", the SP1 State shows "Unknown"? Hope someone here can have some idea. Thanks.
hwittenb:
Quote from: DavidNewton01 on April 27, 2013, 08:01:39 pm
Since nobody replied this topic yet, I would like to make my question simpler: what could cause this situation: in the call status page in OBI100, while the Phone State shows "Connected", the SP1 State shows "Unknown"?
To answer your question, incomplete sip signalling. I assume you are trying to forward the call to a direct ip sip uri on the OBi. The OBi is sloppy when it comes to direct ip calling.
I tried forwarding a CallCentric DID to my OBi110 using direct ip sip uri. I call the DID number. The OBi phone rings. I answer. If you look at Call Status on the OBi it shows the Phone State connected and it shows the SP2 State as unknown. There is no audio.
A wireshark trace shows the incoming Sip INVITE and the OBi responding with a Sip 200 OK when the call is answered. The OBi sends repeated Sip 200 OK requests but the OBi does not receive the necessary an ACK response from CallCentric. In this case there is no audio because the rtp voice stream is not started. Looking at the Sip 200 OK message sent to CallCentric, the Contact: field in the message header contains my local network ip address instead of the external ip address. I believe that is the problem, CallCentric trying to send the ACK to a local network address.
If I change SP2 so that it is registered to the voip provider setup under SP2, and the ITSP Profile setting X_DiscoverPublicAddress is checked (the default) then the OBi puts the external ip address in the Contact: field in the Sip 200 OK message and it all works as it should. It needs both of the settings I mentioned.
In your case you had one-way audio so it is not identical to this case. The problem, though, is probably similiar. Could be a miscommunication of the rtp port to send the rtp stream.
DavidNewton01:
Quote from: hwittenb on April 27, 2013, 09:42:13 pm
I tried forwarding a CallCentric DID to my OBi110 using direct ip sip uri. I call the DID number. The OBi phone rings. I answer. If you look at Call Status on the OBi it shows the Phone State connected and it shows the SP2 State as unknown. There is no audio.
If I change SP2 so that it is registered to the voip provider setup under SP2, and the ITSP Profile setting X_DiscoverPublicAddress is checked (the default) then the OBi puts the external ip address in the Contact: field in the Sip 200 OK message and it all works as it should. It needs both of the settings I mentioned.
Thanks a lot for this important information. Your problem is basically the same as mine. The strange thing is that this setup worked very well before, and then suddenly stopped working. For my case, SP1 is registered to an local asterisk server using local IP. X_discoverPublicAddress is the default (checked). Probably the local register server IP address caused the problem.
hwittenb:
To get a better idea of what is happening you should run the syslog on your OBi. With that you can look at the sip signalling. OBihai has instructions on how to do that and a syslog download link here:
http://www.obitalk.com/forum/index.php?action=printpage;topic=707.0
Navigation
[0] Message Index
[#] Next page