On 05/08/2012 04:24 PM, Kevin P. Fleming wrote:
On 05/08/2012 08:50 AM, Jonathan Rose wrote:----- Original Message -----From: "Jonas Kellens"<jonas.kellens@xxxxxxxxxx>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@xxxxxxxxxxxxxxxx>Sent: Tuesday, May 8, 2012 7:13:30 AM Subject: Asterisk 1.8 Transfer CallerID Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number.That would be because this is the expected behavior. The call isn't coming from the outside caller, it's coming from the person who transferred it.I expect here that colleague B would see the external calling number on the screen of his IP-phone. How can I get this behaviour ? Thanks. Jonas.Getting this behavior shouldn't be too hard I wouldn't think. First, be aware that the Dial command has an option s(x) which is described: s(x): Force the outgoing callerid tag parameter to be set to the string<x>. Works with the f option. So if you simply transfer to a dial application with that option, you can force the callerid to be whatever you want it to be. You can also retrieve the callerid of the original caller and put it on your transferring peer in a variable when starting the call. I'm not exactly sure on the specifics of that right now, but I'm pretty sure it should be possible. So then when you are making the transfer to dial, you just use that variable as your argument to the s option.This is overkill, although it is certainly a way to approach it.If the OP's SIP peers for his phones are configured to send Remote-Party-ID or P-Asserted-Identity to those phones
This is a setting in sip.conf ?
This can be tested by using the CONNECTEDLINE() dialplan function to send anything desired to a phone that is in a call with Asterisk.
According to the wiki it works on Polycom but not on Grandstream.Can you tell me what exactly needs to be supported on the IP-phone ? Is it a certain RFC or something else ?
Jonas. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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