"Once the call is answered, your PSTN gateway shouldn't be involved in the transfer process."
Well, I understand it works like this (actually, I checked with my test system and it does :-). Devices: Snom300, Cisco 2651 ISDn router) 1. PSTN gateway sends an INVITE to target1. 2. Target1 answers with OK to the PSTN gateway and establishes the call 3. Transfer starts. Target1 sends a REFER to the PSTN gateway with target2 as Refer-To. 4. The PSTN gateway INVITEs target2. Since I sent the mail, we made it working with the Cisco. The solution involved a new dial peer matching the short internal numbers. If the destination pattern matches the internal numbering plan, the dial peer sends back the call to SipX. This way the Cisco gateway can handle the internal SipX users when they are targeted with Refer-To. Funny how much the Cisco gateway does not care about domains. I am interested in experiences with SIP trunks. Do I understand correctly that the Ingate device handles REFER if the trunk is not able to? Regards, Gabor ________________________________ From: Tony Graziano [mailto:[email protected]] Sent: 25 February 2009 11:08 To: [email protected]; Gabor Paller Subject: Re: [sipx-users] Best practices for transfer Once the call is answered, your PSTN gateway shouldn't be involved in the transfer process. When a call is transferred it is handled by the proxy, and your UA's (phones, target1/target2) should be able to handle this procedure. If ANY call comes in from the PSTN and hits an AA, and then can transfer to an extension successfully, it means your PSTN gateway handles the refer properly. Once audio is established with a target, you should be able to tarnsfer repeatedly. If it is a siptrunking device, Ingate for example, you should make sure the ITSP supports the feature, and your SBC or siptrunk gateway does as well. I has "subsequent transfer failures" with an Ingate using a siptrunk, but solved it here: http://track.sipfoundry.org/browse/XECS-2098 >>> "Gabor Paller" <[email protected]> 02/25/09 3:58 AM >>> Hi, I am curious about the best practices for transferring external incoming calls. The scenario is very common: Source (an external number) calls target1 (an internal number, SipX user). Target1 transfers the call to target2 (another internal number). This models a common case of a secretary transferring calls. I understand that SIP-wise, the scenario depends on REFER request sent from target1 device to the device representing the source. As the source is a PSTN number, the device representing the source is either a gateway or a SIP trunking equipment which don't know about the target2 internal number. What is the best way of getting the gateway/trunking equipment to know about target2? Regards, Gabor _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
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