caller ID visible in line status but no caller ID in peer number field

[sambgp]

Hi

I have the Obi 110 attached to a GSM FCT (GSM gateway) i can see the caller ID of calls coming in on the phone & line status page of the 110 but the PSTN caller's number is not visible in the peer number field in the call status or call history as a result the calls being carried over SP2 dont carry the PSTN caller ID inspite of "X_spoofcallerid" option being enabled. Would appreciate any suggestion as to why this is happening or how to resolve it?

[ianobi]

If CallerID does not show up in call history, then the OBi has not processed it for some reason. "LastCallerInfo" does show that caller ID exists, but not that OBi has processed it.

Several settings can affect how the OBi110 detects CallerID from the Line Port. To give the OBi more time to detect CallerID try increasing:
Physical Interfaces -> LINE Port -> RingDelay
This might get annoying if increased too much as it delays ringing from incoming calls.

You could also try changing these settings:
Physical Interfaces->LINE Port->Ring Detection -> RingThreshold to 19.35-23.65 and/or RingValidationTime to 150.

Check this is compatable with your GSM device:
Physical Interfaces->LINE Port->Port Settings -> CallerIDDetectMethod

[sambgp]

Thanks for the response, i increased the ring delay from 4 to 5 secs now the caller ID shows in the peer number field. However, would you know how to make the obihai talk to a Linksys SPA 3102 without using any ITSP? I have a random account (not belonging to any VSP) set up on line1 of the SPA 3102, i just want my GSM/PSTN calls to get forwarded to this account.

I tried setting up a similar setup on the Obihai as i have on the SPA 3102 by using a random number in SP2  sip credentials, disabling x-register, and using the following dial plan on Line port SP2(xyz@domain.net:5080), domain.net is just the Dynamic DNS account with which the SPA 3102 is registered and xyz is the sip username.

Problem is that the calls do get forwarded but when the other party using the SPA3102 hangs up it is not detected by the obihai and it continues to keep the line port off-hook

[ianobi]

I have used the setup you describe with various softphones and another OBi, but not with a SPA 3102. I cannot see any reason for the SPA to be different.

If I understand correctly, calls incoming from your PSTN/GSM forward correctly to the SPA, set up correctly and speech is ok. When the SPA user goes on hook, are you sure that the call is holding from the SPA end? It may be the GSM device not clearing down with the Line Port.

The Line Port and GSM device must agree on some of the settings in:
Physical Interfaces > Line Port > PSTN Disconnect Detection

It might be worth taking the SPA out of the setup for testing. Making sure that the GSM device and OBi work correctly with clearing from both directions. Change the Line Port InboundCallRoute back to ph to test to the OBi110 phone.

[sambgp]

Yes calls coming in get properly forwarded to the SPA and the speech is ok, but when the SPA user hangs up the Obihai doesnt get the signal from SPA to hangup. Since the Obi doesnt hangup, the VoIP leg remains connected and consequently the PSTN leg of the call also remains connected. I think in this case the GSM gateway should get a signal through the Line port (FXO  port) that the VoIP side has hung up. Conversely, if the GSM caller hangs up first then OBi detects the polarity reversal and hangs up the VoIP side. Strange thing is that when i register both the SPA and Obi with a common ITSP and make calls through it the call gets properly disconnected from the VoIP side when the SPA user hangs up. Doesnt look like the GSM gateway is the bottleneck here because if it was not clearing and holding up the line, then it would behave in the same fashion even if an ITSP was used.

Would like to mention that there was also a problem with the SPA user calling in on the Obihai SP1 with there being no audio but i followed instructions in this thread http://www.obitalk.com/forum/index.php?topic=4024.0 and put the Dynamic DNS name of my router to which Obihai is connected as the outbound proxy in the SPA which has resolved the audio problem

[ianobi]

Seems odd. If the SIP signalling works to set up the call, then I'm guessing it should work when the SPA sends the "bye" signal to the OBi.

I would try to eliminate the GSM device or the SPA from the setup and see if the problem goes away. Any chance of setting up an incoming call from sp1 on the OBi and forwarding it to the SPA, then seeing how call clearing works? To eliminate the SPA you might set up a softphone on a PC/laptop with the same settings and forward an incoming call via the GSM device to it. Phonerlite is free and has a good debugging feature to trace back through SIP messages.

[sambgp]

Yes its very odd and frustrating, inspite of having the caller-ID info available i cant forward it to the VoIP side unless i use an ITSP because of this disconnection problem.

I tried forwarding calls from SP1(which is registered with betamax) to the SPA instead of SP2 but the problem persists, the moment i use direct calling with URI's it stops disconnecting from the VoIP side.

Will try to eliminate the GSM device as you have suggested and connect the obi to my landline to see if the problem goes away. If nothing works out will have to resort to debugging, will keep this thread updated.