[asterisk-users] Asterisk 1.8 broken MWI

[satish patel]

Following is my debug and look like its not sending MWI NOTIFY message to phone

Reliably Transmitting (no NAT) to 172.30.245.143:5060:
OPTIONS sip:7623 at 172.30.245.143 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.30.1.46>;tag=as26352734
To: <sip:7623 at 172.30.245.143>
Contact: <sip:asterisk at 172.30.1.46:5060>
Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c at 172.30.1.46:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-branch-1.8-r321926
Date: Wed, 08 Jun 2011 14:49:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.30.245.143:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3
From: "asterisk" <sip:asterisk at 172.30.1.46>;tag=as26352734
To: <sip:7623 at 172.30.245.143>;tag=E777D3B9-F605D562
CSeq: 102 OPTIONS
Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c at 172.30.1.46:5060
Contact: <sip:7623 at 172.30.245.143>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '44c3ac7b4c37786c3fda41e12d1a907c at 172.30.1.46:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 172.30.245.143:5060:
OPTIONS sip:7623 at 172.30.245.143 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.30.1.46>;tag=as0c8778f4
To: <sip:7623 at 172.30.245.143>
Contact: <sip:asterisk at 172.30.1.46:5060>
Call-ID: 50d5cc4d5510ae014c6641702faea18d at 172.30.1.46:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-branch-1.8-r321926
Date: Wed, 08 Jun 2011 14:50:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.30.245.143:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37
From: "asterisk" <sip:asterisk at 172.30.1.46>;tag=as0c8778f4
To: <sip:7623 at 172.30.245.143>;tag=47557FCE-869CEA2F
CSeq: 102 OPTIONS
Call-ID: 50d5cc4d5510ae014c6641702faea18d at 172.30.1.46:5060
Contact: <sip:7623 at 172.30.245.143>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---


From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Wed, 8 Jun 2011 14:43:57 +0000
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI








Truly speaking, I went though that file and i found nothing in that file related major changes.  It was working perfect before 1.2 

May be i am missing some configuration option. Do you know any debug method to make it work ?

> From: EWieling at nyigc.com
> To: asterisk-users at lists.digium.com
> Date: Wed, 8 Jun 2011 10:34:16 -0400
> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
> 
> All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball.
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> > satish patel
> > Sent: Wednesday, June 08, 2011 9:57 AM
> > To: asterisk-users
> > Subject: [asterisk-users] Asterisk 1.8 broken MWI
> >
> > Hi ALL,
> >
> > After upgrade 1.8 my MWI wasn't working I do have setting in
> > voicemail.conf.  Do i need to do anything else to fix my MWI
> > on polycom 501 ? It was working with 1.2 asterisk.
> >
> > pollmailboxes=yes
> >
> >
> 
> --
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