Olivier
2008-Aug-01 07:57 UTC
[asterisk-users] Comparing origination from CLI and from AMI
Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123 192.168.64.5 -> Context: local 192.168.64.5 -> Priority: 1 I must add both 9122 and 9123 extensions are SIP extensions which default to "local" context. When using B (AMI/FOP), I've got a : -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.100.195 > Channel SIP/9122-081d8f68 was never answered. where 192.168.100.195 is SIP/9122 hardphone IP address When using A (CLI), everything works ok. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20080801/83638101/attachment.htm
Olivier
2008-Aug-05 05:42 UTC
[asterisk-users] Comparing origination from CLI and from AMI
Hi, A closer look showed that SIP FOP-Originated calls are "self-addressed" While some phones tolerate that, others reply with 480 moved temporarily. Case 1: Command Line Interface with Thomson hardphone After I typed "originate SIP/9122 application dial Local/9123 at local", 1st SIP message received is an INVITE from server like this: INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2e7491fc;rport From: "asterisk" <sip:asterisk at 192.168.100.254<sip%3Aasterisk at 192.168.100.254>>;tag=as72b7dcafTo: <sip:9121 at 192.168.100.198:5060;user=phone> Contact: <sip:asterisk at 192.168.100.254 <sip%3Aasterisk at 192.168.100.254>> Call-ID: 7abbcf4b377fd55e2390f48b2fde320c at 192.168.100.254 CSeq: 102 INVITE User-Agent: Asterisk PBX My SIP extension 9121 Thomson hardphone starts to ring and everything is fine. Case 2: Drag and drop origination with FOP and Thomson hardphone After I dragged 9121 icon into 9123 icon, I got this : INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport From: "9121 Guest1" <sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>>;tag=as237a9159To: <sip:9121 at 192.168.100.198:5060;user=phone> Contact: <sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>> Call-ID: 6bddeb200c2aee553856dab4098c6f8e at 192.168.100.254 CSeq: 102 INVITE User-Agent: Asterisk PBX then this : SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport From: "9121 Guest1"<sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>>;tag=as237a9159To: <sip:9121 at 192.168.100.198:5060;user=phone>;tag=c0a80101-a611e Call-ID: 6bddeb200c2aee553856dab4098c6f8e at 192.168.100.254 CSeq: 102 INVITE Content-Length: 0 With this, my SIP extension 9121 Thomson hardphone didn't start to ring. Case 3: Drag and drop origination with FOP and Siemens Gigaset S45 hardphone After I dragged 7531 icon into 9123 icon, I got this : INVITE sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2a4a8fb1;rport From: "7531 Marcelo Dup" <sip:7531 at 192.168.100.254<sip%3A7531 at 192.168.100.254>>;tag=as5c8e7711To: <sip:7531 at 192.168.100.197:5060> Contact: <sip:7531 at 192.168.100.254 <sip%3A7531 at 192.168.100.254>> Call-ID: 027f21ae2248de196334494155885ceb at 192.168.100.254 CSeq: 102 INVITE User-Agent: Asterisk PBX then this : SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2a4a8fb1;rport=5060 From: "7531 Marcelo Dup" <sip:7531 at 192.168.100.254<sip%3A7531 at 192.168.100.254>>;tag=as5c8e7711To: <sip:7531 at 192.168.100.197:5060>;tag=4240967763 Call-ID: 027f21ae2248de196334494155885ceb at 192.168.100.254 CSeq: 102 INVITE Contact: "Herv?" <sip:7531 at 192.168.100.197:5060> Content-Length: 0 My SIP extension 7531 Siemens Gigaset S45 hardphone starts to ring and everything is fine. So, bottom line is : - AMI/FOP and CLI do not generate the same behaviour : with AMI/FOP, first INVITE comes from <sip:callerextension at serveraddress> while with CLI, it comes from <sip:asterisk at serveraddress> - some SIP phones accept <sip:callerextension at serveraddress> INVITE messages while others don't. Do you agree with this conclusion ? Which workaround would you try ? Regards 2008/8/1 Olivier <oza-4h07 at myamail.com>> Hi, > > Using FOP, I've met a situation which makes me ask this simple question : > > Are both A and B commands bellow equivalent ? > > A. CLI: > originate SIP/9122 application dial Local/9123 at local > > B. AMI/FOP: > 192.168.64.5 -> Action: Originate > 192.168.64.5 -> Channel: SIP/9122 > 192.168.64.5 -> Async: True > 192.168.64.5 -> Callerid: 9122 Guest2 <9122> > 192.168.64.5 -> Exten: 9123 > 192.168.64.5 -> Context: local > 192.168.64.5 -> Priority: 1 > > > I must add both 9122 and 9123 extensions are SIP extensions which default > to "local" context. > > When using B (AMI/FOP), I've got a : > -- Got SIP response 480 "Temporarily Unavailable" back from > 192.168.100.195 > > Channel SIP/9122-081d8f68 was never answered. > where 192.168.100.195 is SIP/9122 hardphone IP address > > When using A (CLI), everything works ok. > > Regards >-------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20080805/bbfc39c4/attachment.htm
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