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HP Recommended

I've followed the FAQ section on how to enable the BLF with Asterisk. I am using Asterisk 11.0.1 (and tried 11.1.0-rc1) and two Polycom 650 4.0.3F.

 

Asterisk sends multiple SIP NOTIFY messages.

 

First:

NOTIFY sip:phone-01@10.23.12.40 SIP/2.0
Via: SIP/2.0/UDP 10.23.6.150:5060;branch=z9hG4bK79a5f30b;rport
Max-Forwards: 70
From: <sip:1202@sip.reucon.net>;tag=as497340c7
To: "Stefan Reuter" <sip:phone-01@sip.reucon.net>;tag=BE1C780C-C4DB7E0D
Contact: <sip:1202@10.23.6.150:5060>
Call-ID: dfaed1ab-1ca5b34-ab43b3f5@10.23.12.40
CSeq: 271 NOTIFY
User-Agent: Asterisk PBX 11.0.1
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 571

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="169" state="full" entity="sip:1202@sip.reucon.net">
<dialog id="1202" call-id="pickup-dfaed1ab-1ca5b34-ab43b3f5@10.23.12.40" local-tag="BE1C780C-C4DB7E0D" remote-tag="as497340c7" direction="recipient">
<remote>
<identity display="1202">sip:1202@sip.reucon.net</identity>
<target uri="sip:1202@sip.reucon.net"/>
</remote>
<local>
<identity display="1202">sip:1202@sip.reucon.net</identity>
<target uri="sip:1202@sip.reucon.net"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

 Second:

NOTIFY sip:phone-01@10.23.12.40 SIP/2.0
Via: SIP/2.0/UDP 10.23.6.150:5060;branch=z9hG4bK6edcb3a8;rport
Max-Forwards: 70
From: <sip:1202@sip.reucon.net>;tag=as497340c7
To: "Stefan Reuter" <sip:phone-01@sip.reucon.net>;tag=BE1C780C-C4DB7E0D
Contact: <sip:1202@10.23.6.150:5060>
Call-ID: dfaed1ab-1ca5b34-ab43b3f5@10.23.12.40
CSeq: 272 NOTIFY
User-Agent: Asterisk PBX 11.0.1
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 584

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="170" state="full" entity="sip:1202@sip.reucon.net">
<dialog id="1202" call-id="pickup-dfaed1ab-1ca5b34-ab43b3f5@10.23.12.40" local-tag="BE1C780C-C4DB7E0D" remote-tag="as497340c7" direction="recipient">
<remote>
<identity display="Stefan Reuter">sip:1201@sip.reucon.net</identity>
<target uri="sip:1201@sip.reucon.net"/>
</remote>
<local>
<identity display="phone-02">sip:1202@sip.reucon.net</identity>
<target uri="sip:1202@sip.reucon.net"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

 Despite the second NOTIFY containing the correct Caller IDs the Polycom continues to show the data from the first message, i.e. To: 1202(1202) From: 1202(1202) while I would expect them to be updated to To: phone-02(1202) From: Stefan Reuter(1201).

 

Is this an issue with Asterisk or Polycom? Any fixes or work arounds?

3 REPLIES 3
HP Recommended

Hello srt,

welcome to the Polycom Community.

What kind of scenario is this ?

 

I did the FAQ post as I tested this on my Asterisk 1.6 / 1.8

 

My Call flow would be a simple monitoring of lines for incoming calls. Is you scenario different where the monitored line has a number displayed that would change during the signaling? 

 

In order to work this correctly I suggest you raise a ticket via your Polycom reseller with Polycom support so we can verify this with our developers once a wireshark trace and the phone configuration has been checked.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
HP Recommended

My scenario is just like yours: two phones monitoring each other for incoming calls.

 

The issue seems to be introduced in Asterisk after 1.8 with the fix for this issue: https://issues.asterisk.org/jira/browse/ASTERISK-16735. This fix seems to work fine with SNOM phones which update the UI if they receveive updated caller ids but doesn't work for Polycom phones. As you see in my comment on ASTERISK-16735 I've patched my Asterisk server now to not send the first SIP NOTIFY message as a workaround.

 

It would be great if you could pass that on to engineering to see if updated caller ids is something you want to support. Please understand that I prefer not to go through the hassle to use support through my reseller. If you had a more open support process that would make things a lot easier.

HP Recommended

Hi Stefan,

 

thanks for the clarification.

 

Does this follow an RFC or is this a specific Asterisk issue ?

 

Best Regards

 

Steffen Baier

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
† The opinions expressed above are the personal opinions of the authors, not of HP. By using this site, you accept the <a href="https://www8.hp.com/us/en/terms-of-use.html" class="udrlinesmall">Terms of Use</a> and <a href="/t5/custom/page/page-id/hp.rulespage" class="udrlinesmall"> Rules of Participation</a>.