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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
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Olivier

2008-Aug-05 05:42 UTC

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[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
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