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@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@172.30.1.46>;tag=as26352734 To: <sip:7623@172.30.245.143> Contact: <sip:asterisk@172.30.1.46:5060> Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@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@172.30.1.46>;tag=as26352734 To: <sip:7623@172.30.245.143>;tag=E777D3B9-F605D562 CSeq: 102 OPTIONS Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060 Contact: <sip:7623@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@172.30.1.46:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 172.30.245.143:5060: OPTIONS sip:7623@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@172.30.1.46>;tag=as0c8778f4 To: <sip:7623@172.30.245.143> Contact: <sip:asterisk@172.30.1.46:5060> Call-ID: 50d5cc4d5510ae014c6641702faea18d@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@172.30.1.46>;tag=as0c8778f4 To: <sip:7623@172.30.245.143>;tag=47557FCE-869CEA2F CSeq: 102 OPTIONS Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060 Contact: <sip:7623@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...@hotmail.com To: asterisk-users@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: ewiel...@nyigc.com > To: asterisk-users@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-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@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 > > > > > > -- > _____________________________________________________________________ > -- 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 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
