Chappo Posted April 19, 2013 Report Share Posted April 19, 2013 Guys, As the title states I am having some issues with how the internal phones register the incoming calls. The phones on an attended transfer or even picking up an external trunk directly will register either the star code or the person who performed the attended transfer. Is there a way to override this on the pbx side of things? callrecord_received_local0!: 45@pbx.domain.com.au callrecord_received_local1!: 45@pbx.domain.com.au callrecord_received_local2!: 45@pbx.domain.com.au callrecord_received_local3!: 45@pbx.domain.com.au callrecord_received_local4!: 45@pbx.domain.com.au callrecord_received_remote0!: sip:40@pbx.domain.com.au callrecord_received_remote1!: sip:47@pbx.domain.com.au callrecord_received_remote2!: sip:40@pbx.domain.com.au callrecord_received_remote3!: sip:61@pbx.domain.com.au callrecord_received_remote4!: sip:*6011273@pbx.domain.com.au Trunk Settings: Request-URI Let the System Decide From: Based on trunk account info To: Same as request-URI P-Asserted-Identity: Don't use Header P-Preferred-Identity: Don't use Header Remote-Party-ID: Don't use Header Privacy Indication: Other: id The problem I have is modifying any of these values results in a 502 Bad Gateway error as shown below: [5] 2013/04/19 12:04:41: SIP Rx udp:125.213.160.81:5060: SIP/2.0 502 Bad Gateway Via: SIP/2.0/UDP 172.28.1.5:5060;branch=z9hG4bK-a6f55ac37452f8b6bbc0f28e35179a25;rport To: <sip:947180000@sip00.mynetfone.com.au> From: "JR"<sip:OurPhoneNo@pbx.domain.com.au;user=phone>;tag=13990 Call-ID: 08b383d5@pbx CSeq: 5001 INVITE User-Agent: ENSR3.0.66.41-IS1 Content-Length: 0 [5] 2013/04/19 12:04:41: SIP Tx udp:125.213.160.81:5060: ACK sip:947180000@sip00.mynetfone.com.au;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.1.5:5060;branch=z9hG4bK-a6f55ac37452f8b6bbc0f28e35179a25;rport From: "JR" <sip:OurPhoneNo@pbx.domain.com.au;user=phone>;tag=13990 To: <sip:947180000@sip00.mynetfone.com.au> Call-ID: 08b383d5@pbx CSeq: 5001 ACK Max-Forwards: 70 Contact: <sip:09406157@172.28.1.5:5060;transport=udp> Privacy: id Content-Length: 0 [5] 2013/04/19 12:04:41: INVITE Response 502 Bad Gateway: Terminate 08b383d5@pbx [5] 2013/04/19 12:04:41: SIP Tx tls:172.28.1.84:3629: SIP/2.0 502 Bad Gateway Via: SIP/2.0/TLS 172.28.1.84:3629;branch=z9hG4bK-66fvzg68v8or;rport=3629 From: "JR" <sip:48@pbx.domain.com.au>;tag=9q3f1xp4xm To: <sip:947180000@pbx.domain.com.au;user=phone>;tag=a7fad5b5fd Call-ID: 49c270511180-6ninip39wyuz CSeq: 1 INVITE Contact: <sip:48@172.28.1.5:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.8 Quote Link to comment Share on other sites More sharing options...
Chappo Posted May 1, 2013 Author Report Share Posted May 1, 2013 Anyone? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 1, 2013 Report Share Posted May 1, 2013 Well the Bad Gateway does come from the service provider (ENSR3.0.66.41-IS1). Probably you have to tweak the SIP trunk header settings to make it work. If the provider has a description where he wants the authentication info and where the caller-ID info, it should speed up the process. Quote Link to comment Share on other sites More sharing options...
Chappo Posted May 3, 2013 Author Report Share Posted May 3, 2013 This is what has been provided by the VSP. INVITE sip:1300887899@sip10.mynetfone.com.au:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.150:5060;branch=z9hG4bK-d8754z-591caa70b8520754-1---d8754z-;rport=61973;received=xxx.xxx.xxx.xxx Max-Forwards: 70 Contact: <sip:09123456@xxx.xxx.xxx.xxx:5060> To: <sip:1300887899@sip10.mynetfone.com.au:5060> From: "0280088000"<sip: 09123456@sip10.mynetfone.com.au:5060>;tag=e53ba10d Call-ID: MTY4YTMxOTE2MDk4ZTAzMThkYmFlY2QwY2MxZTE4MDE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: Content-Length: 332 Remote-Party-ID: "0280088000"<sip: 0280088000@sip10.mynetfone.com.au:5060>;party=calling ^ Please ensure that your remote party ID or P asserted identity string is exactly the same as the "from" string – The above example would fail because it is not the same. Typically the P Asserted Identity would be your internal extension number OR your DID. Quote Link to comment Share on other sites More sharing options...
Vodia support Posted May 3, 2013 Report Share Posted May 3, 2013 You can choose other under the P-asserted identity and put something like <sip:{0280088000}@{sip10.mynetfone.com.au};user=phone> or <sip:{0280088000}@{trunk-outbound-proxy};user=phone> not sure if 0280088000 is the DID provided to you Quote Link to comment Share on other sites More sharing options...
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