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How to originate calls with SIP hardphones not accepting "self-addressed" calls


  • From: "Safet Susic" <panoramisk@xxxxxxxxx>
  • Date: Tue, 5 Aug 2008 19:37:35 +0200
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Hi,

I can originate calls with FOP with some phones while I can't do the same
with other phones.
I could focus that root cause is that some SIP phones accept what I call
"self-address" while other don't.

I made 3 trials :

Case 1: Command Line Interface with Thomson ST2030 hardphone

After I typed "originate SIP/9122 application dial Local/9123@xxxxx", 1st
SIP message received is an INVITE from server like this:

INVITE sip:9121@xxxxxxxxxxxxxxx:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2e7491fc;rport
From: "asterisk" <sip:asterisk@xxxxxxxxxxxxxxx<sip%3Aasterisk@xxxxxxxxxxxxxxx>
;tag=as72b7dcaf
To: <sip:9121@xxxxxxxxxxxxxxx:5060;user=phone>
Contact: <sip:asterisk@xxxxxxxxxxxxxxx <sip%3Aasterisk@xxxxxxxxxxxxxxx>>
Call-ID: 7abbcf4b377fd55e2390f48b2fde320c@xxxxxxxxxxxxxxx
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@xxxxxxxxxxxxxxx:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport
From: "9121 Guest1" <sip:9121@xxxxxxxxxxxxxxx <sip%3A9121@xxxxxxxxxxxxxxx>
;tag=as237a9159
To: <sip:9121@xxxxxxxxxxxxxxx:5060;user=phone>
Contact: <sip:9121@xxxxxxxxxxxxxxx <sip%3A9121@xxxxxxxxxxxxxxx>>
Call-ID: 6bddeb200c2aee553856dab4098c6f8e@xxxxxxxxxxxxxxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX

then my SIP phone replies 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@xxxxxxxxxxxxxxx <sip%3A9121@xxxxxxxxxxxxxxx>
;tag=as237a9159
To: <sip:9121@xxxxxxxxxxxxxxx:5060;user=phone>;tag=c0a80101-a611e
Call-ID: 6bddeb200c2aee553856dab4098c6f8e@xxxxxxxxxxxxxxx
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@xxxxxxxxxxxxxxx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK2a4a8fb1;rport
From: "7531 Marcelo Dup" <sip:7531@xxxxxxxxxxxxxxx<sip%3A7531@xxxxxxxxxxxxxxx>
;tag=as5c8e7711
To: <sip:7531@xxxxxxxxxxxxxxx:5060>
Contact: <sip:7531@xxxxxxxxxxxxxxx <sip%3A7531@xxxxxxxxxxxxxxx>>
Call-ID: 027f21ae2248de196334494155885ceb@xxxxxxxxxxxxxxx
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@xxxxxxxxxxxxxxx<sip%3A7531@xxxxxxxxxxxxxxx>
;tag=as5c8e7711
To: <sip:7531@xxxxxxxxxxxxxxx:5060>;tag=4240967763
Call-ID: 027f21ae2248de196334494155885ceb@xxxxxxxxxxxxxxx
CSeq: 102 INVITE
Contact: "Hervé" <sip:7531@xxxxxxxxxxxxxxx:5060>
Content-Length: 0

My SIP extension 7531 Siemens Gigaset S45 hardphone starts to ring and
everything is fine.




How can I work around this ?
Changing context in op_buttons.cfg , I couldn't keep first leg of call to
send INVITE with "From: <sip:9121@xxxxxxxxxxxxxxx<sip%3A9121@xxxxxxxxxxxxxxx>>"
address.

How would you proceed ?

Regards

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