Re: [asterisk-users] Dialing a SIP URI with an ;ext= parameter

2014-01-21 Thread Leandro Dardini
I am going to try a Lync server/asterisk integration, so I really
appreciate!

Leandro


2014/1/21 Lincoln King-Cliby linc...@controlworks.com

 Ok, so now I just feel kind of stupid. After I got home I decided to play
 with this a little more.



 After far too long I realized that part of the issue was Asterisk parsing
 the ; as a beginning of a comment (hindsight=duh).

 A little bit more experimenting and (though I could swear I tried this
 before) replacing the ; with \; works.



 That is, to dial a E.164 normalized number with an extension configured as
 tel:+14404491100;ext=1407 +14404491100;ext=1407 with the SIP Peer for
 the Lync mediation server named “lync” the working dial() is



 Dial(SIP/lync/+14404491100\;ext=1407)



 Hope this may save someone else time down the road.



 --

 Lincoln King-Cliby, CTS, DMC-D

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lincoln King-Cliby
 *Sent:* Monday, January 20, 2014 5:04 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Dialing a SIP URI with an ;ext= parameter



 Hi All,



 In the midst of trying to pilot a deployment of Microsoft Lync (mainly for
 non-voice collaboration, specifically IM) and integrate it with our
 Asterisk (11.6.0 if it matters) deployment and a “everything in one place”
 tool when people are out of the office.



 I have everything on the voice side playing  nice from the Lync side
 (Lync-Lync, Lync-Asterisk, Lync-Asterisk-PSTN)  but I can’t get calls
 from Asterisk-Lync passing.



 I think the root issue is Lync demands that the “line URI” be entered in a
 E.164 normalized format, and further specifies that if an extension is
 specified it should be entered as ;ext=. So, e.g. when I have myself set up
 in LYNC my Line URI is entered as 
 “tel:+144044911100;ext=1407+144044911100;ext=1407”.




 If I try feeding that into an Asterisk DIAL() using any format I can think
 of (specific examples below) the call fails and the following is logged to
 console; it looks like Asterisk is dropping the “;ext=”…

   == Using SIP RTP CoS mark 5

 -- Executing [1407@yyy:1] Dial(xx, SIP/lync/
 +14404491100) in new stack

   == Using SIP RTP CoS mark 5

 -- Called SIP/lync/+14404491100

 -- Got SIP response 485 Ambiguous back from IP address and port of
 Lync mediation server

   == Everyone is busy/congested at this time (1:0/0/1)

 -- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL'



 On the other hand, if I change my line URI to a “random” and unused in
 Lync E.164 number without an extension and change the DIAL() to reflect
 that number… the call succeeds, so it seems like I’ve narrowed it down to
 just needing to figure out how to properly pass the extension to Lync.



 The Googling I turned up didn’t seem too positive (and suggested using an
 Exchange Unified Messaging auto attendant and forcing the user to redial
 the extension once connected to the AA was the only alternative for non-DID
 users) but it seems like it should be relatively simple to bridge (what
 seems like a very small) gap.



 Here are the least embarrassing variations on Dial I’ve tried



 Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(“SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(SIP/lync/+14404491100/1407) -- call ‘sits there’ and multiple
 “sip_xmit of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1:
 Invalid argument” logged to console





 Any assistance, is as always very appreciated.



 Thanks!



 Lincoln







 --

 Lincoln King-Cliby, CTS, DMC-D

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



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Re: [asterisk-users] Dialing a SIP URI with an ;ext= parameter

2014-01-20 Thread Lincoln King-Cliby
Ok, so now I just feel kind of stupid. After I got home I decided to play with 
this a little more.

After far too long I realized that part of the issue was Asterisk parsing the ; 
as a beginning of a comment (hindsight=duh).
A little bit more experimenting and (though I could swear I tried this before) 
replacing the ; with \; works.

That is, to dial a E.164 normalized number with an extension configured as 
tel:+14404491100;ext=1407 with the SIP Peer for the Lync mediation server named 
lync the working dial() is

Dial(SIP/lync/+14404491100\;ext=1407)

Hope this may save someone else time down the road.

--
Lincoln King-Cliby, CTS, DMC-D
Commercial Market Director
Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
V: 440.449.1100 x1107 F: 440-449-1106 I: 
http://www.controlworks.comhttp://www.controlworks.com/
Crestron Services Provider

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby
Sent: Monday, January 20, 2014 5:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialing a SIP URI with an ;ext= parameter

Hi All,

In the midst of trying to pilot a deployment of Microsoft Lync (mainly for 
non-voice collaboration, specifically IM) and integrate it with our Asterisk 
(11.6.0 if it matters) deployment and a everything in one place tool when 
people are out of the office.

I have everything on the voice side playing  nice from the Lync side 
(Lync-Lync, Lync-Asterisk, Lync-Asterisk-PSTN)  but I can't get calls from 
Asterisk-Lync passing.

I think the root issue is Lync demands that the line URI be entered in a 
E.164 normalized format, and further specifies that if an extension is 
specified it should be entered as ;ext=. So, e.g. when I have myself set up in 
LYNC my Line URI is entered as tel:+144044911100;ext=1407.

If I try feeding that into an Asterisk DIAL() using any format I can think of 
(specific examples below) the call fails and the following is logged to 
console; it looks like Asterisk is dropping the ;ext=...
  == Using SIP RTP CoS mark 5
-- Executing [1407@yyy:1] Dial(xx, SIP/lync/+14404491100) 
in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/lync/+14404491100
-- Got SIP response 485 Ambiguous back from IP address and port of Lync 
mediation server
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL'

On the other hand, if I change my line URI to a random and unused in Lync 
E.164 number without an extension and change the DIAL() to reflect that 
number... the call succeeds, so it seems like I've narrowed it down to just 
needing to figure out how to properly pass the extension to Lync.

The Googling I turned up didn't seem too positive (and suggested using an 
Exchange Unified Messaging auto attendant and forcing the user to redial the 
extension once connected to the AA was the only alternative for non-DID users) 
but it seems like it should be relatively simple to bridge (what seems like a 
very small) gap.

Here are the least embarrassing variations on Dial I've tried

Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above
Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above
Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above
Dial(SIP/lync/+14404491100/1407) -- call 'sits there' and multiple sip_xmit 
of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1: Invalid argument 
logged to console


Any assistance, is as always very appreciated.

Thanks!

Lincoln



--
Lincoln King-Cliby, CTS, DMC-D
Commercial Market Director
Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
V: 440.449.1100 x1107 F: 440-449-1106 I: 
http://www.controlworks.comhttp://www.controlworks.com/
Crestron Services Provider

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users