<div dir="ltr">Hi,<br><br>A closer look showed that SIP FOP-Originated calls are "self-addressed"<br>While some phones tolerate that, others reply with 480 moved temporarily.<br><br><br>Case 1: Command Line Interface with Thomson hardphone<br>
<br>After I typed "originate SIP/9122 application dial Local/9123@local", 1st SIP message received is an INVITE from server like this:<br><br>INVITE sip:9121@192.168.100.198:5060;user=phone SIP/2.0<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2e7491fc;rport<br>
From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.100.254">sip:asterisk@192.168.100.254</a>>;tag=as72b7dcaf<br>To: <sip:9121@192.168.100.198:5060;user=phone><br>Contact: <<a href="mailto:sip%3Aasterisk@192.168.100.254">sip:asterisk@192.168.100.254</a>><br>
Call-ID: <a href="mailto:7abbcf4b377fd55e2390f48b2fde320c@192.168.100.254">7abbcf4b377fd55e2390f48b2fde320c@192.168.100.254</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br><br>My SIP extension 9121 Thomson hardphone starts to ring and everything is fine.<br>
<br><br>Case 2: Drag and drop origination with FOP and Thomson hardphone<br>
<br>After I dragged 9121 icon into 9123 icon, I got this :<br><br>INVITE sip:9121@192.168.100.198:5060;user=phone SIP/2.0<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport<br>From: "9121 Guest1" <<a href="mailto:sip%3A9121@192.168.100.254">sip:9121@192.168.100.254</a>>;tag=as237a9159<br>
To: <sip:9121@192.168.100.198:5060;user=phone><br>Contact: <<a href="mailto:sip%3A9121@192.168.100.254">sip:9121@192.168.100.254</a>><br>Call-ID: <a href="mailto:6bddeb200c2aee553856dab4098c6f8e@192.168.100.254">6bddeb200c2aee553856dab4098c6f8e@192.168.100.254</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br><br>then this :<br><br>SIP/2.0 480 Temporarily Unavailable<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport<br>From: "9121 Guest1"<<a href="mailto:sip%3A9121@192.168.100.254">sip:9121@192.168.100.254</a>>;tag=as237a9159<br>
To: <sip:9121@192.168.100.198:5060;user=phone>;tag=c0a80101-a611e<br>Call-ID: <a href="mailto:6bddeb200c2aee553856dab4098c6f8e@192.168.100.254">6bddeb200c2aee553856dab4098c6f8e@192.168.100.254</a><br>CSeq: 102 INVITE<br>
Content-Length: 0<br><br>With this, my SIP extension 9121 Thomson hardphone didn't start to ring.<br><br><br>Case 3: Drag and drop origination with FOP and Siemens Gigaset S45 hardphone<br>
<br>After I dragged 7531 icon into 9123 icon, I got this :<br><br><br>INVITE <a href="http://sip:7531@192.168.100.197:5060">sip:7531@192.168.100.197:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2a4a8fb1;rport<br>
From: "7531 Marcelo Dup" <<a href="mailto:sip%3A7531@192.168.100.254">sip:7531@192.168.100.254</a>>;tag=as5c8e7711<br>To: <<a href="http://sip:7531@192.168.100.197:5060">sip:7531@192.168.100.197:5060</a>><br>
Contact: <<a href="mailto:sip%3A7531@192.168.100.254">sip:7531@192.168.100.254</a>><br>Call-ID: <a href="mailto:027f21ae2248de196334494155885ceb@192.168.100.254">027f21ae2248de196334494155885ceb@192.168.100.254</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br><br>then this :<br><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2a4a8fb1;rport=5060<br>From: "7531 Marcelo Dup" <<a href="mailto:sip%3A7531@192.168.100.254">sip:7531@192.168.100.254</a>>;tag=as5c8e7711<br>
To: <<a href="http://sip:7531@192.168.100.197:5060">sip:7531@192.168.100.197:5060</a>>;tag=4240967763<br>Call-ID: <a href="mailto:027f21ae2248de196334494155885ceb@192.168.100.254">027f21ae2248de196334494155885ceb@192.168.100.254</a><br>
CSeq: 102 INVITE<br>Contact: "Hervé" <<a href="http://sip:7531@192.168.100.197:5060">sip:7531@192.168.100.197:5060</a>><br>Content-Length: 0<br><br>My SIP extension 7531 Siemens Gigaset S45 hardphone starts to ring and everything is fine.<br>
<br><br><br>So, bottom line is :<br>- AMI/FOP and CLI do not generate the same behaviour :<br> with AMI/FOP, first INVITE comes from <sip:callerextension@serveraddress> while with CLI, it comes from <sip:asterisk@serveraddress><br>
- some SIP phones accept <sip:callerextension@serveraddress> INVITE messages while others don't.<br><br><br>Do you agree with this conclusion ?<br>Which workaround would you try ?<br><br>Regards<br><br><br><div class="gmail_quote">
2008/8/1 Olivier <span dir="ltr"><<a href="mailto:oza-4h07@myamail.com" target="_blank">oza-4h07@myamail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">Hi,<br><br>Using FOP, I've met a situation which makes me ask this simple question :<br>
<br>Are both A and B commands bellow equivalent ?<br><br>A. CLI:<br>originate SIP/9122 application dial Local/9123@local<br>
<br>B. AMI/FOP:<br><a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Action: Originate<br><a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Channel: SIP/9122<br><a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Async: True<br>
<a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Callerid: 9122 Guest2 <9122><br><a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Exten: 9123<br><a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Context: local<br>
<a href="http://192.168.64.5" target="_blank">192.168.64.5</a> -> Priority: 1<br><br><br>I must add both 9122 and 9123 extensions are SIP extensions which default to "local" context.<br><br>When using B (AMI/FOP), I've got a :<br>
-- Got SIP response 480 "Temporarily Unavailable" back from <a href="http://192.168.100.195" target="_blank">192.168.100.195</a><br> > Channel SIP/9122-081d8f68 was never answered.<br>where <a href="http://192.168.100.195" target="_blank">192.168.100.195</a> is SIP/9122 hardphone IP address<br>
<br>When using A (CLI), everything works ok.<br><br>Regards<br></div>
</blockquote></div><br></div>