FORWARDING OR TRANSFERING SIP CALLS

[canflyguy]

After attempting a few different setups and either answering a call on the phone and transferring it and also trying to have the voicemail transfer a call through to another number, it appears there is no audio passthrough between the incoming SIP trunk and the outgoing SIP trunk on the conference. The call routes and connects, but no audio passthrough. Is it possible or impossible? Also caller ID is sent from the BCM on the conference call made by the set, but no caller ID when the voicemail transfers the call? I did put a public OLI on the main voicemail DN.

Anyone got this working?

 

[cndr99]

What system are you using, and what kind of sets do you have?

Verulam Telephone:
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Serving Eastern Ontario

 

[canflyguy]

System is a Nortel/Avaya BCM. Sets don't matter 7316 or 1140's. Unless you meant what voip service? I did play around before with findme/followme working and presenting the incoming caller ID with audio, but that was using an analog trunk and then going out on a SIP trunk.

 

[cndr99]

I asked about the sets, because I noticed a peculiarity when I was experimenting with a BCM400 R3.7: VOIP calls from one kind of set to another (ie 7316->1140, Avaya 9611G-> i2050, etc) would have no audio unless you put the call on hold briefly and then picked it up again. I assumed that lack of documentation for the local SIP server abandonware I was using, and improper codec settings or provisioning were the issue, but never got it resolved.



Verulam Telephone:
Communicating is our Pleasure!

Serving Eastern Ontario

 

[exsmogger]

As far as the CLID issue, check under Resources - IPTrunks - General - IP Trunk Settings and see if "Forward redirected OLI" is set to "Last Redirect". That's what fixed the forwarded caller ID for me.

Brian Cox
Georgia Telephone

http://Georgia-Telephone.com

http://www.linkedin.com/in/briancox1952

 

[DocRobotnik]

I always wondering what OLI mean, i tought it was the former extention if you rename it.

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[curlycord]

Originating Line Identification"

And I had same issue, same fix too.

OLI used on Sets to show your DID or any number you wish in the DID range when calling out, also is some cases must be programmed to allow the call to go out.



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Toronto, Canada

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[curlycord]

Any luck on this one?

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Toronto, Canada

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[curlycord]

Try this as I was plaing with a clients site, same issue.

General under “Scope” section from “NONE ”to “RTP Keepalives” and parameters of that: Initial Keepalives = Enabled, I left the Timeout at default 0.

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Toronto, Canada

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[canflyguy]

I'll do a test on a system I have here. One of my sites does transfer calls out and seems to have no issue, so perhaps I can see what his settings are and get back to here with results.

 

[curlycord]

Thanks!!

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Toronto, Canada

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[canflyguy]

I resolved the issue here at the test location by "Enable local NAT compensation". I have since been into my router here which is on Bell Fibe and found a feature that was turned on which I have since disabled called SIP ALG. My understanding is that it causes issues. I have now to go back and test with the "Enable local NAT compensation" turned off to see if it is no longer required although it did make the audio flow through instantly previously.

 

[exsmogger]

SIP ALG causes more problems than it solves. Turning on NAT pinhole maintenance can also be helpful if you have problems with incoming SIP trunk calls.

Brian Cox
Georgia Telephone

http://Georgia-Telephone.com

http://www.linkedin.com/in/briancox1952