MessageWell after thinking a little more about your scenario I have had situations where I have hairpinned calls back to the device they have come in on and in my experience without canreinvite=no the device will usually complain that a loop was detected. In situations were I am using SER(SIP EXPRESS ROUTER) I have fixed the problem by rewriting the headed to trick the device out of the loop detect scenario. Anyway in theory if the 7206 doesn't detect a loop if you have the sip.conf setup to the canreinvite=yes then the audio channel should not be proxied by *.
In this calling multiple channels how do you deal with the cellphone voicemail not answering the call immediately if the phone is off or out of range which wouldn't allow for the opportunity for the other extension to be answered.
----- Original Message -----
From: B. J. Bomar
To: asterisk-users at lists.digium.com
Sent: Thursday, February 05, 2004 1:37 PM
Subject: RE: [Asterisk-Users] Release phone call
The way we have it setup is simply calling multiple numbers/channels. It is either setup manually in the configs, or through a very ugly menu interface I constructed.
B. J.
-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Glenn Dalgliesh
Sent: Thursday, February 05, 2004 12:31
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Release phone call
I don't really have a answer for you on you issue but have a question about what "find-me" is. I see it on the feature list but am unable to find any real information about it. Is this simply call forward or is their more to it.
thanks
----- Original Message -----
From: B. J. Bomar
To: asterisk-users at lists.digium.com
Sent: Thursday, February 05, 2004 1:01 PM
Subject: [Asterisk-Users] Release phone call
Hello all, I am trying to figure out how to have * release a phone call. We are noticing some call quality issues on people who have a "find-me" feature, and answer the call through a cell phone. Here is the call path we are seeing, and all VoIP connections are using SIP.
PSTN ---> Cisco 7206 ---> * Server
^-----------| ^-------------|
Hopefully the diagram makes sense, but in case it doesn't, let me try to explain. A call comes in from PSTN into our Cisco7206 with PRI card. It then goes to our * server, which then forwards the call back through the Cisco to a cell phone on PSTN. I am wanting to have * release the call to the Cisco once the call is connected. Any thoughts or ideas?
Thanks.
B. J.
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