kelvin Posted August 25, 2008 Report Posted August 25, 2008 hi Support, i am make use of tel alias as global alias for inter branch call, so that branch extension able contact each others using this global alias but after upgrade to version 3.0.0.2998. It stop working, everytime call global alias pbx return not found message. kindly advice the solution. thx regards, kelvin Quote
asterisk_nicht_mehr Posted August 25, 2008 Report Posted August 25, 2008 Did you confirm in the documentation that that is how one is supposed to use the Global? Quote
Vodia PBX Posted August 25, 2008 Report Posted August 25, 2008 Did you confirm in the documentation that that is how one is supposed to use the Global? We had to change a couple of things with the alias. The calls must now be routed through a trunk to the other domain. The old way is not suitable for server farms where you have no idea on which server a domain (or tel:-alias) is physically located. As we catch up with the documentation we'll elaborate that on the Wiki in further detail. Quote
kelvin Posted August 26, 2008 Author Report Posted August 26, 2008 We had to change a couple of things with the alias. The calls must now be routed through a trunk to the other domain. The old way is not suitable for server farms where you have no idea on which server a domain (or tel:-alias) is physically located. As we catch up with the documentation we'll elaborate that on the Wiki in further detail. for this case, what alternate way i can use to route inter branch calls. thx regards, kelvin Quote
kelvin Posted August 28, 2008 Author Report Posted August 28, 2008 for this case, what alternate way i can use to route inter branch calls. thx regards, kelvin hi support, any update on above request? thx regards, kelvin Quote
Vodia PBX Posted August 28, 2008 Report Posted August 28, 2008 hi support, any update on above request? thx regards, kelvin Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one. Quote
kelvin Posted September 4, 2008 Author Report Posted September 4, 2008 Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one. hi support, i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx SIP/2.0 183 Ringing Via: SIP/2.0/TLS 192.168.1.176:3101;branch=z9hG4bK-2v9nief449ot;rport=3101 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=dkm152o0i5 To: <sip:365215@pbx.pgcomms.com.my;user=phone>;tag=539287bafe Call-ID: 3c3a1fd5d921-1d8euqo6pq07 CSeq: 1 INVITE Contact: <sip:102@192.168.1.10:5081;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pgcomms-PBX/3.0.0.2998 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 433 v=0 o=- 18443 18443 IN IP4 192.168.1.10 s=- c=IN IP4 192.168.1.10 t=0 0 m=audio 54216 RTP/AVP 18 3 2 0 8 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iIE4YWtoEmL5VNDncRdIeA2SKR2GKZly23Qc+k83 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: INVITE sip:65215@127.0.0.1:5080;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone> Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Max-Forwards: 70 Contact: <sip:102@127.0.0.1:5080;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pgcomms-PBX/3.0.0.2998 P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my> Content-Type: application/sdp Content-Length: 331 v=0 o=- 20980 20980 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 58324 RTP/AVP 18 3 2 0 8 9 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [5] 2008/09/04 17:26:40: Received loopback request without tag [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57660 [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57661 [5] 2008/09/04 17:26:40: Identify trunk (IP address/port match) 23 [9] 2008/09/04 17:26:40: Resolve 17492: aaaa udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17492: a udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17492: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Content-Length: 0 [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Content-Length: 0 [9] 2008/09/04 17:26:40: Resolve 17493: aaaa udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17493: a udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17493: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE User-Agent: pgcomms-PBX/3.0.0.2998 WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5 Content-Length: 0 [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE User-Agent: pgcomms-PBX/3.0.0.2998 WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5 Content-Length: 0 [7] 2008/09/04 17:26:40: Call 7b24e046@pbx#15046: Clear last INVITE [9] 2008/09/04 17:26:40: Resolve 17494: url sip:127.0.0.1:5080 [9] 2008/09/04 17:26:40: Resolve 17494: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: ACK sip:65215@127.0.0.1:5080;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 ACK Max-Forwards: 70 Contact: <sip:102@127.0.0.1:5080;transport=udp> P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my> Content-Length: 0 Quote
Vodia PBX Posted September 4, 2008 Report Posted September 4, 2008 hi support, i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx .... try to use a ANI for the extension that wants to call the other name or a Prefix on the trunk. The PBX probably thinks the call comes from an extension, not from a trunk. Quote
kelvin Posted September 5, 2008 Author Report Posted September 5, 2008 .... try to use a ANI for the extension that wants to call the other name or a Prefix on the trunk. The PBX probably thinks the call comes from an extension, not from a trunk. by default the ANI = caller extension no right? i want to display Caller extension so that we can identify the calls. by the way what is this message mean? "[5] 2008/09/05 14:06:22: Received loopback request without tag" Quote
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