Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
Hi Marek, Thank you - I figured out my issue, which was that the MWI subscribes to a PJSIP AOR, which in turns monitors a mailbox, not directly an actual mailbox. Mike -Original Message- From: asterisk-users On Behalf Of Marek Greško Sent: November 19, 2021 03:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't Hello Michael, I was also struggling with solicited MWI after moving to pjsip. My problem was I was defining mailbox=111@extensioncontext. But the correct context in the mailbox command is to be defined by context in voicemail.conf. My voicemails were all defined in the context default (see voicemail.conf) and the mailbox command should look like this: mailbox=111@default. Hope this helps. I do not know whether this is also your problem. Marek 2021-11-14 16:38 GMT+01:00, Mike : > Hi, > > > > Just recently moved over from chan_sip to PJSIP and am slowly > cleaning up whatever needs to be. > > > > I can't seem to make sollicitated MWI work, but unsollicitated works fine. > > > > > I got my phones subscribing to mailbox@context (i.e. 100@whatever) > > > > I have my related AOR entry (realtime, in a DB) set to > mailboxes=100@whatever . I can see it is set properly by using the > command "pjsip show aor " > > > > But when I turn pjsip logger on, I see messages from the phones > subscribing and SIP/2.0 401 Unauthorized messages back. > > > > If I put the same column in my realtime DB (mailboxes) for ENPOINT to > the same value (100@whatever) then it works fine, MWI works on the phone. > > > > For a few reasons I'd like to get MWI working in sollicitated mode > instead. Is there a trick to it? > > > > I upgraded to Asterisk 18.8.0 just to see if a later patch fixed > anything, so I am current. > > > > > > > > > > > > > > Michael > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
Hello Michael, I was also struggling with solicited MWI after moving to pjsip. My problem was I was defining mailbox=111@extensioncontext. But the correct context in the mailbox command is to be defined by context in voicemail.conf. My voicemails were all defined in the context default (see voicemail.conf) and the mailbox command should look like this: mailbox=111@default. Hope this helps. I do not know whether this is also your problem. Marek 2021-11-14 16:38 GMT+01:00, Mike : > Hi, > > > > Just recently moved over from chan_sip to PJSIP and am slowly cleaning up > whatever needs to be. > > > > I can't seem to make sollicitated MWI work, but unsollicitated works fine. > > > > > I got my phones subscribing to mailbox@context (i.e. 100@whatever) > > > > I have my related AOR entry (realtime, in a DB) set to > mailboxes=100@whatever . I can see it is set properly by using the command > "pjsip show aor " > > > > But when I turn pjsip logger on, I see messages from the phones > subscribing and SIP/2.0 401 Unauthorized messages back. > > > > If I put the same column in my realtime DB (mailboxes) for ENPOINT to the > same value (100@whatever) then it works fine, MWI works on the phone. > > > > For a few reasons I'd like to get MWI working in sollicitated mode > instead. Is there a trick to it? > > > > I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything, > so I am current. > > > > > > > > > > > > > > Michael > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
On Mon, Nov 15, 2021 at 8:34 PM Mike wrote: > So I think I am halfway there. > > > > It seems configuring 100@whatever in the aor turns the MWI subscription > from a 401 unauthorized into a 404 not found. > > > > So I’m guessing the MWI subscribe goes through, since the aor now allows > it, but then fails when asterisk actually looks for the mailbox once passes > the “security” of mailboxes=100@whatever. > > > > The thing is, the mailbox is only in a table but asterisk definitely sees > it (and saves msg with no issues). “Voicemail show users for whatever” > lists it as being there. > > > > But the mailbox is neither in voicemail.conf nor users.conf (by design). > Is this needed? > > > > Is there a better place to ask this sort of question? > You'd need to actually show the SIP SUBSCRIBE and show the AOR. It states what is being subscribed to, and if it's not subscribing to the same name as the AOR then a 404 would be sent. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
So I think I am halfway there. It seems configuring 100@whatever in the aor turns the MWI subscription from a 401 unauthorized into a 404 not found. So I'm guessing the MWI subscribe goes through, since the aor now allows it, but then fails when asterisk actually looks for the mailbox once passes the "security" of mailboxes=100@whatever. The thing is, the mailbox is only in a table but asterisk definitely sees it (and saves msg with no issues). "Voicemail show users for whatever" lists it as being there. But the mailbox is neither in voicemail.conf nor users.conf (by design). Is this needed? Is there a better place to ask this sort of question? From: asterisk-users On Behalf Of Mike Sent: November 14, 2021 10:38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't Hi, Just recently moved over from chan_sip to PJSIP and am slowly cleaning up whatever needs to be. I can't seem to make sollicitated MWI work, but unsollicitated works fine. I got my phones subscribing to mailbox@context (i.e. 100@whatever) I have my related AOR entry (realtime, in a DB) set to mailboxes=100@whatever . I can see it is set properly by using the command "pjsip show aor " But when I turn pjsip logger on, I see messages from the phones subscribing and SIP/2.0 401 Unauthorized messages back. If I put the same column in my realtime DB (mailboxes) for ENPOINT to the same value (100@whatever) then it works fine, MWI works on the phone. For a few reasons I'd like to get MWI working in sollicitated mode instead. Is there a trick to it? I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything, so I am current. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
Hi, Just recently moved over from chan_sip to PJSIP and am slowly cleaning up whatever needs to be. I can't seem to make sollicitated MWI work, but unsollicitated works fine. I got my phones subscribing to mailbox@context (i.e. 100@whatever) I have my related AOR entry (realtime, in a DB) set to mailboxes=100@whatever . I can see it is set properly by using the command "pjsip show aor " But when I turn pjsip logger on, I see messages from the phones subscribing and SIP/2.0 401 Unauthorized messages back. If I put the same column in my realtime DB (mailboxes) for ENPOINT to the same value (100@whatever) then it works fine, MWI works on the phone. For a few reasons I'd like to get MWI working in sollicitated mode instead. Is there a trick to it? I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything, so I am current. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI Delayed on Polycom VVX phones
>>> I'll try that patch later on today. I'm not using the mailboxes=##, but >>> will try the patch just the same. Patch applied and fixed my problem, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI Delayed on Polycom VVX phones
>>> https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17 >>> I am using "mailboxes=##@default" and had the issue as well (before). >>> Michael Thanks Michael! I'll try that patch later on today. I'm not using the mailboxes=##, but will try the patch just the same. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI Delayed on Polycom VVX phones
> Am 15.01.2019 um 15:23 schrieb Doug Lytle : > > Hi all, > > When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has > resulted in a MWI clearing delay of around 5 minutes. > > After listening to a voicemail and deleting it, the Polycom VVX 601's MWI > light is left on for around five minutes, before clearing. > > Installing Asterisk 13.24.1 did not fix this. > > Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in > the change logs for 13.24.0 > > [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead > of mailboxes=##@default > > Any suggestions on what to look at to diagnose? > > Doug Hi Doug, applying this patch helped in my case (with AstLinux 1.3.x + Asterisk 13.24.1): https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17 I am using "mailboxes=##@default" and had the issue as well (before). Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI Delayed on Polycom VVX phones
Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in the change logs for 13.24.0 [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default Any suggestions on what to look at to diagnose? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI and PJSIP
Does something change with MWI when moving from SIP to PJSIP? Seems my phone isn't be alerted of its new VM. -- _ -- 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
Re: [asterisk-users] MWI issue
I have a situation that I need help with. I have 2 phone systems, 1 Asterisk and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes into an extension that the Asterisk server owns, I re-direct it to a different number that is owned by the Avaya System. If that Avaya extension does not answer it, I send it to the voicemail on the Avaya Messaging system for the extension that it came in on the Asterisk box. Once that happens, I need to send a MWI indicator to an application on the desktop of the Avaya User that there is a voicemail for that mailbox. I see the SIP Notify come in from Avaya for the extension (I did this with a tcpdump). My question is how do I configure Asterisk to act on that request and call an agi program to do what I want. Any help would be appreciated. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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
[asterisk-users] MWI publish VIA pjsip for non sip channels
Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work with PJSIP? The net effect here is that I want to get away from res_xmpp, if possible. Matt Hoskins | NPG Corp | Systems Architect -- _ -- 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
Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
Matt Hoskins wrote: Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? Yes. SIP is simply used as the transport mechanism. It works pretty much the same as res_xmpp except without needing an XMPP server. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
Of course, I left out a detail that may (or may not change) the answer. I'm using the external chan-sccp-b sccp module, not the chan_skinny bundled with asterisk. Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Thursday, October 30, 2014 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels Matt Hoskins wrote: Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? Yes. SIP is simply used as the transport mechanism. It works pretty much the same as res_xmpp except without needing an XMPP server. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29 tr=YmFzZQ%3D%3D http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5 vcmc%3Dr=YmFzZQ%3D%3D -- _ -- Bandwidth and Colocation Provided by http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hcGktZGlnaXR hbC5jb20%3Dr=YmFzZQ%3D%3D -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5 vcmcvaGVsbG8%3Dr=YmFzZQ%3D%3D asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL2xpc3RzLmRpZ2l1bS5 jb20vbWFpbG1hbi9saXN0aW5mby9hc3Rlcmlzay11c2Vycw%3D%3Dr=YmFzZQ%3D%3D -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 01N9y9aRM) is spam: Spam: http://spamaway.npgco.com/canit/b.php?i=01N9y9aRMm=82ea4601cf34t=2014103 0c=s Not spam: http://spamaway.npgco.com/canit/b.php?i=01N9y9aRMm=82ea4601cf34t=2014103 0c=n Forget vote: http://spamaway.npgco.com/canit/b.php?i=01N9y9aRMm=82ea4601cf34t=2014103 0c=f -- END-ANTISPAM-VOTING-LINKS -- _ -- 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
Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
Matt Hoskins wrote: Of course, I left out a detail that may (or may not change) the answer. I'm using the external chan-sccp-b sccp module, not the chan_skinny bundled with asterisk. Still doesn't matter. Provided it works with res_xmpp it'll work with the new SIP method. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
Awesome - Thanks for the quick replies. I'm sure I could have tried-and-see but with going from Asterisk 11 to 13, there'd be so many things changing - it helps to know from the outset. Thanks again. Matt Hoskins | NPG Corp | Systems Architect -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Thursday, October 30, 2014 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels Matt Hoskins wrote: Of course, I left out a detail that may (or may not change) the answer. I'm using the external chan-sccp-b sccp module, not the chan_skinny bundled with asterisk. Still doesn't matter. Provided it works with res_xmpp it'll work with the new SIP method. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29 tr=YmFzZQ%3D%3D http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5 vcmc%3Dr=YmFzZQ%3D%3D -- _ -- Bandwidth and Colocation Provided by http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hcGktZGlnaXR hbC5jb20%3Dr=YmFzZQ%3D%3D -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5 vcmcvaGVsbG8%3Dr=YmFzZQ%3D%3D asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL2xpc3RzLmRpZ2l1bS5 jb20vbWFpbG1hbi9saXN0aW5mby9hc3Rlcmlzay11c2Vycw%3D%3Dr=YmFzZQ%3D%3D -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 01N9yiS6F) is spam: Spam: http://spamaway.npgco.com/canit/b.php?i=01N9yiS6Fm=50dc54beaae5t=2014103 0c=s Not spam: http://spamaway.npgco.com/canit/b.php?i=01N9yiS6Fm=50dc54beaae5t=2014103 0c=n Forget vote: http://spamaway.npgco.com/canit/b.php?i=01N9yiS6Fm=50dc54beaae5t=2014103 0c=f -- END-ANTISPAM-VOTING-LINKS -- _ -- 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
[asterisk-users] MWI issue
On Asterisk 1.8.18.0, when the Voice message is transferred from one extension to another extension, MWI on Linksys SPA 508G does work all the time. Somehow sometimes it seems to be working and sometimes it's not. Please advice. Thanks! -- _ -- 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
[asterisk-users] MWI not working - Asterisk 1.8.9.2
Hiii, I am testing MWI on my grandstream and bria. Following is sip show peer 1001 * Name : 1001 Secret : Set MD5Secret: Not set Remote Secret: Not set Context : EXT_1001 Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1001 VM Extension : default LastMsgsSent : 32767/65535 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : 1001 1001 MaxCallBR: 512 kbps Expire : 40 Insecure : no Force rport : No ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : Yes . Following is sip show subscription Peer User Call ID Extension Last state TypeMailboxExpiry 172.16.26.1711002 627149977@172.1 -- none mwi 1002 60 172.16.26.1271001 2068510560-4266 -- none mwi 1001 60 Following is show voicemail - i changed format of the same for general use Mbox User NewMsg 1002 1002 6 1003 1003 0 1004 1004 0 1005 1005 0 1001 1001 28 I can receive listen and also do all stuff using voicemailmain application. But no MWI on any client. is there any thing else i need to check ? can any one help to solve the problem Thanks in advance, Bharat Lalcheta -- _ -- 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
[asterisk-users] MWI for non-subscribed Realtime peers?
Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706] type=friend host=xxx.xxx.xxx.xxx (IP of proxy) port=5060 nat=no mailbox=1234@customer subscribemwi=no defaultuser=az5134939706 Every time a voicemail has been left in the mailbox 1234@customer, a NOTIFY is sent off to the proxy. Remember, the peer doesn't register or send SUBSCRIBE to Asterisk, but subscribemwi=no forces NOTIFY to be sent anyway. However, I am struggling to get the same thing working for Realtime peers. I have rtcachefriends=yes set in sip.conf. But I never see the peer loaded from database and no NOTIFY is ever sent. Is it possible to user Realtime this way? What will trigger loading of the peer? Best regards, Jan Blom -- _ -- 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
Re: [asterisk-users] MWI for non-subscribed Realtime peers?
Let me answer my own question. That may save someone's frustration in the
future.
The problem is that the Realtime peer never gets loaded, since SIP REGISTER and
SIP SUBSCRIBE never reaches Asterisk.
Doing a simple sip show peer az5134939706 load from CLI will force load of
peer. However, I needed a way of having this done automatically on startup for
all (many!) peers. A number of methods are suggested by people (use Google) but
they all seemed like hacks to me.
Finally I realized, after rereading chan_sip.c, the solution was to force load
the peer from dialplan. If I do this just before I send a caller to voicemail,
I can be sure the peer is available when MWI NOTIFY should be sent. Just add
this to the dialplan:
same = n,NoOp(${SIPPEER(az5134939706)})
Good luck with your Realtime MWI hacking!
Best regards,
Jan
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Blom
Sent: den 23 november 2011 13:04
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MWI for non-subscribed Realtime peers?
Hi,
I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations
and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to
the proxy even when the SUBSCRIBE haven't been received. I can configure a user
in sip.conf that works:
[az5134939706]
type=friend
host=xxx.xxx.xxx.xxx (IP of proxy)
port=5060
nat=no
mailbox=1234@customer
subscribemwi=no
defaultuser=az5134939706
Every time a voicemail has been left in the mailbox 1234@customer, a NOTIFY is
sent off to the proxy. Remember, the peer doesn't register or send SUBSCRIBE to
Asterisk, but subscribemwi=no forces NOTIFY to be sent anyway.
However, I am struggling to get the same thing working for Realtime peers. I
have rtcachefriends=yes set in sip.conf. But I never see the peer loaded from
database and no NOTIFY is ever sent.
Is it possible to user Realtime this way? What will trigger loading of the peer?
Best regards,
Jan Blom
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Re: [asterisk-users] MWI not working on most ATAs in Asterisk 1.6.2.17
maill...@lightspeed.ca writes: We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. My testing of 1.6.2.17 as well as the svn branch of 1.6.2 a few weeks ago indicated that MWI was fairly broken. I managed to get it working somewhat reasonably on Snom phones with a combination of subscribemwi=no (despite the fact that the Snom phones subscribe for MWI!?), pollmailboxes=yes and pollfreq=30 (despite the fact that we have nothing but Asterisk touching the voicemail files). In the testing, I managed to get Asterisk to flood the phones with tens or maybe hundreds of MWI's by simply leaving one voice mail. No, I have not had time to file bugs, we simply did a fall back to 1.6.0.28 + a lot of patches. It was not the most serious bug anyway; the larger problem is that Asterisk deadlocks. /Benny -- _ -- 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
[asterisk-users] MWI not working on most ATAs in Asterisk 1.6.2.17
We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. While not exhaustive, these are the ATAs that don't work: Linksys SPA2102 Linksys PAP2T-3.1.15 Thomson 780 Thomson 784 Unfortunately, this doesn't bode well for us, since these represent the vast majority of our customers. I did manage to find one ATA that does work without a hitch on the new server, a Sipura SPA2002. Which of course, is discontinued. The Thomsons coincidentally, never worked on our old server either, but they're here because if someone knows how to make them work, we'd be very happy. I know that by default, Asterisk will send SIP NOTIFY messages to the ATA to inform it of waiting messages, and I see those notify messages being sent, but somehow something goes wrong from there. Any information about further troubleshooting this problem will be very welcome. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] MWI SUBSCRIBE Settings
Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox...@from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug we get: --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Creating new subscription Sending to 10.0.0.4 : 5090 (no NAT) list_route: hop: sip:2...@10.0.0.4:5090 Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866 Content-Length: 0 --- SIP read from UDP:10.0.0.4:5090 --- SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: sip:2...@10.0.0.4:5090 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 - --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 --- Transmitting (no NAT) to 10.0.0.4:5090 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: sip:2...@10.0.0.10;tag=6d8c6ac6 To: sip:2...@10.0.0.10;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=ePBX, nonce=26c43866
[asterisk-users] MWI Assistance
Hi, I'm struggling to get the MWI set up on a few Polycom phones. The setup is like this. I've got a few phones in the context called [company2_phones] and I've got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf (i'm actually using sip realtime if that makes a difference), i've entered mailbo...@company2 (1 being the name of the mailbox) However, the phone doesnt subscribe to the mailbox status. In the Polycom documentation, it asks me to provide:- ASCII encoded string containing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL (6416 or 6...@polycom.com) But I have no idea what to enter. I've tried everything I can think of but I get this in the Asterisk CLI:- [2010-10-11 23:06:08] NOTICE[18424]: chan_sip.c:16331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: company2_201 company2_201 is the user part listed in sip.conf for that particular extension. What do I enter in order to get it to request the mailbox status? Any assistance would be appreciated. Thanks Dan -- _ -- 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
Re: [asterisk-users] MWI Assistance
I'm struggling to get the MWI set up on a few Polycom phones. Sorted. From voip-info. http://www.voip-info.org/wiki/view/Asterisk+RealTime The database peers/users are not kept in memory. These are only loaded when we have a call and then deleted, so there's no support for NAT keep-alives (qualify=) or voicemail indications for these peers. NOTE: If you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers' - see rtcachefriends=yes. The downside to this is that if you change anything in the database, you need to do a 'sip reload' (for major changes) or 'sip prune realtime PEERNAME' (for single peer changes) before they become active. -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
Hi Dave, Thought I'd give you an update - I completely rebuilt my astdb the other night by renaming it, having * recreate it and then re-creating all my custom entries in it. Didn't have any effect, I had somebody report false MWI notifications again earlier this morning. -- Matt On Tue, Mar 9, 2010 at 11:34 PM, Matt Watson m...@mattgwatson.ca wrote: Hi Dave, Sure enough my astdb does contain references to VM files as shown with strings - doing the database dump however does not show the references. I'm not sure about the internals of how Berk DB works, however I;m also seeing references to lots of other data that really shouldn't be part of my config anymore either - like I can see some employee names that are no longer a part of our company and thus have been deleted from our * config, some several years ago. I suspect that berkdb is just not overwriting some of the data for whatever reason and has some internal mechanism for knowing what to ignore. I believe I can probably test your theory tomorrow evening though, I don't think I have too much in my astdb that can't be easily re-created, I think I can probably delete my astdb entirely and regenerate it. I'll just need to take a closer look at it first though. I would however like to believe that if * is no longer supposed to be using berkdb for any VM reference data, that any calls to read the voicemail counts from the DB should have been removed. -- Matt On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.comwrote: So a couple of questions I have for you Matt... If you run strings on your astdb file are you seeing references to messages files in it? #strings /var/lib/asterisk/astdb | grep -i msg and if so... If you run a db_dump185 on your astdb file do the references go away? #db_dump185 -p -f /tmp/astdb.dump astdb -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
On 8 Mar 2010, at 22:08, Dave Poirier wrote: Top posting to remain consistent... I drop litter because everyone else does. ;) W -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
Will Payne wrote: On 8 Mar 2010, at 22:08, Dave Poirier wrote: Top posting to remain consistent... I drop litter because everyone else does. ;) W Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read by being needlessly pedantic. It's like those people who decide that, even though traffic is moving along at an average of 70mph, they're going to drive 55 in the fast lane to 'teach everyone the proper speed.' They're statistically MORE likely to cause accidents (or, in LA, get shot) than those travelling along with traffic at a speed above the posted speed limit. On some positions, it is not helpful to be unwavering. N. -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
On 9 Mar 2010, at 11:47, SIP wrote: Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read by being needlessly pedantic. it just seemed like a 'I know this is wrong, but...' comment :) Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you snip the quote down to the relevant portion, you can reply where you like, regardless of what's gone on beforehand. (Surely there's no such thing as 'needlessly' pedantic - all pedantry is necessary :) W -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
Will Payne wrote: it just seemed like a 'I know this is wrong, but...' comment :) Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you snip the quote down to the relevant portion, you can reply where you like, regardless of what's gone on beforehand. (Surely there's no such thing as 'needlessly' pedantic - all pedantry is necessary :) W Unless it's errant. Then you upset Churchill. N. -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
Hi Dave, Sure enough my astdb does contain references to VM files as shown with strings - doing the database dump however does not show the references. I'm not sure about the internals of how Berk DB works, however I;m also seeing references to lots of other data that really shouldn't be part of my config anymore either - like I can see some employee names that are no longer a part of our company and thus have been deleted from our * config, some several years ago. I suspect that berkdb is just not overwriting some of the data for whatever reason and has some internal mechanism for knowing what to ignore. I believe I can probably test your theory tomorrow evening though, I don't think I have too much in my astdb that can't be easily re-created, I think I can probably delete my astdb entirely and regenerate it. I'll just need to take a closer look at it first though. I would however like to believe that if * is no longer supposed to be using berkdb for any VM reference data, that any calls to read the voicemail counts from the DB should have been removed. -- Matt On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.com wrote: So a couple of questions I have for you Matt... If you run strings on your astdb file are you seeing references to messages files in it? #strings /var/lib/asterisk/astdb | grep -i msg and if so... If you run a db_dump185 on your astdb file do the references go away? #db_dump185 -p -f /tmp/astdb.dump astdb -- _ -- 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
Re: [asterisk-users] MWI and 1.6.1
Top posting to remain consistent... Matt, After doing a little more digging I'm starting to think that it has something to do with our upgrade from 1.4 to 1.6 (nothing like stating the obvious). I'm now suspecting that astdb has something to do with it. I ran strings on astdb which lives in /var/lib/asterisk/ and found references to messages files inside the DB. Maybe they lived there in version 1.4 but not in 1.6. Somebody more clueful than me can probably clarify on this. Doing a dump of the same database doesn't show the references to those files. Could this be what is generating the MWI events? So my thought was to just dump the database in a good state and reload. Unfortunately trying to load the dump leaves me with a newer version of the database files that Asterisk recognize. Asterisk uses a very old version 1 of Berkeley DB. Anyone know how to load a db_load as version 1.85/1.86? I can't seem to figure it out. So a couple of questions I have for you Matt... If you run strings on your astdb file are you seeing references to messages files in it? #strings /var/lib/asterisk/astdb | grep -i msg and if so... If you run a db_dump185 on your astdb file do the references go away? #db_dump185 -p -f /tmp/astdb.dump astdb Then look at the resulting file in /tmp and see if any references to message files are there. At this point I'm completely shooting in the dark but It's the best I've come up with. Let me know your results and perhaps we can figure this out. Thanks, Dave On Thu, Mar 4, 2010 at 4:40 PM, Matt Watson m...@mattgwatson.ca wrote: I'm having this EXACT same problem, I haven;t been able to narrow down the cause of it yet, but it seems to me that users are receiving notifications for voicemails in mailboxes that belong to other people, as sometimes their mail count magically disappears, which I have been suspecting is when somebody else checks their VM. I found the problem also exists in 1.6.2 which is where I first noticed it (upgraded from 1.4.x to 1.6.2.x). I tried downgrading to 1.6.1 and the problem seemed not quite as bad, but I know its still present. I was actually quite surprised to find that nobody had previously mentioned the problem on this list when I came across it so I thought it might of been something specific to my situation. Even if you turn the polling options back on in the voicemail conf file the problem still persists. We are using all Aastra phones - a mix of 9133i, 9112i, 480, 35i, 57i phones - but the problem seem unrelated to the make/model of the phone based on seeing you having the same problem with Polycom's. Not sure that it should matter, but we are using FreePBX 2.6 ontop of asterisk and running it in users and devices mode (as apposed to the default extensions mode). If you do a voicemail show users from the Asterisk console it shows the correct VM counts for the mailboxes, so its not that Asterisk is counting them incorrectly, it just seems to be sending the notifications of VMs to the wrong places. I'm suddenly very glad I;m not alone on this one! I;m more than happy to do any testing of patches if anybody has any suggestions. -- Matt On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier dpoir...@mesd.k12.or.uswrote: We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for messages, the system will tell them that they have none and the light goes out. I know that starting in 1.6 Asterisk moved from a polling system to an event based system but it's unclear to me what is causing these events to be generated. Anyone else experience this? Any tips, suggestions? Thanks, Dave -- _ -- 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:
Re: [asterisk-users] MWI and 1.6.1
I'm having this EXACT same problem, I haven;t been able to narrow down the cause of it yet, but it seems to me that users are receiving notifications for voicemails in mailboxes that belong to other people, as sometimes their mail count magically disappears, which I have been suspecting is when somebody else checks their VM. I found the problem also exists in 1.6.2 which is where I first noticed it (upgraded from 1.4.x to 1.6.2.x). I tried downgrading to 1.6.1 and the problem seemed not quite as bad, but I know its still present. I was actually quite surprised to find that nobody had previously mentioned the problem on this list when I came across it so I thought it might of been something specific to my situation. Even if you turn the polling options back on in the voicemail conf file the problem still persists. We are using all Aastra phones - a mix of 9133i, 9112i, 480, 35i, 57i phones - but the problem seem unrelated to the make/model of the phone based on seeing you having the same problem with Polycom's. Not sure that it should matter, but we are using FreePBX 2.6 ontop of asterisk and running it in users and devices mode (as apposed to the default extensions mode). If you do a voicemail show users from the Asterisk console it shows the correct VM counts for the mailboxes, so its not that Asterisk is counting them incorrectly, it just seems to be sending the notifications of VMs to the wrong places. I'm suddenly very glad I;m not alone on this one! I;m more than happy to do any testing of patches if anybody has any suggestions. -- Matt On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier dpoir...@mesd.k12.or.uswrote: We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for messages, the system will tell them that they have none and the light goes out. I know that starting in 1.6 Asterisk moved from a polling system to an event based system but it's unclear to me what is causing these events to be generated. Anyone else experience this? Any tips, suggestions? Thanks, Dave -- _ -- 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
[asterisk-users] MWI and 1.6.1
We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for messages, the system will tell them that they have none and the light goes out. I know that starting in 1.6 Asterisk moved from a polling system to an event based system but it's unclear to me what is causing these events to be generated. Anyone else experience this? Any tips, suggestions? Thanks, Dave -- _ -- 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
Re: [asterisk-users] MWI count wrong when using IMAP and VM
any ideas ? or file a bug ? Best Regards, - --[ UxBoD ]-- ux...@splatnix.net wrote: | as soon as I delete the two messages I receive in the console :- | | [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP | Warning: Unknown message data: 1 EXPUNGE | [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP | Warning: Unknown message data: 1 EXPUNGE | | Best Regards, | | | - --[ UxBoD ]-- ux...@splatnix.net wrote: | | | [r...@voip ~]# asterisk -V | | Asterisk 1.6.1.11 | | | | When using the above version with IMAP VoiceMail integration when I | | leave a message my SNOM360 it shows 2 message waiting; yet when | | running voicemail show users from the Asterisk CLI it correctly | | reports 1. | | | | It would appear that when the VM is temporarily stored, and the VM | is | | delivered by IMAP to the remote mail account, the MWI is being | | initiated with a incorrect count. | | | | I then delete the VM from either 1) the phone 2) the mail account | the | | MWI goes blank and the message count shows 0 correctly. | | | | I am still trying to debug but any thoughts on this ? | | | | Here is how I have voicemail.conf :- | | | | [general] | | format=wav49 | | maxsecs=180 | | minsecs=5 | | skipms=3000 | | maxsilence=3 | | silencethreshold=128 | | maxlogins=3 | | imapserver=imap_server | | imapfolder=VoiceMail Office | | imapport=993 | | imapflags=ssl | | authuser=imap_user | | authpassword=imap_password | | | | [voicemail] | | 1001 = 1234,user,,,imapuser=u...@imap_server | | | | Best Regards, | | | | | | -- | | This message has been scanned for viruses and | | dangerous content and is believed to be clean. | | | | SplatNIX IT Services :: Innovation through collaboration | | | | | | ___ | | -- Bandwidth and Colocation Provided by http://www.api-digital.com | -- | | | | asterisk-users mailing list | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI count wrong when using IMAP and VM
[r...@voip ~]# asterisk -V Asterisk 1.6.1.11 When using the above version with IMAP VoiceMail integration when I leave a message my SNOM360 it shows 2 message waiting; yet when running voicemail show users from the Asterisk CLI it correctly reports 1. It would appear that when the VM is temporarily stored, and the VM is delivered by IMAP to the remote mail account, the MWI is being initiated with a incorrect count. I then delete the VM from either 1) the phone 2) the mail account the MWI goes blank and the message count shows 0 correctly. I am still trying to debug but any thoughts on this ? Here is how I have voicemail.conf :- [general] format=wav49 maxsecs=180 minsecs=5 skipms=3000 maxsilence=3 silencethreshold=128 maxlogins=3 imapserver=imap_server imapfolder=VoiceMail Office imapport=993 imapflags=ssl authuser=imap_user authpassword=imap_password [voicemail] 1001 = 1234,user,,,imapuser=u...@imap_server Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI count wrong when using IMAP and VM
Following up on this if I leave a second message then the WMI count goes to 4. When I check the voicemail directory on the server I see :- [r...@voip 1001]# ls -lR .: total 20 drwxr-xr-x 2 root root 4096 Dec 4 17:49 INBOX drwxr-xr-x 2 root root 4096 Oct 8 21:02 Old drwxr-xr-x 2 root root 4096 May 13 2009 temp drwxr-xr-x 2 root root 4096 Dec 4 17:49 tmp drwxr-xr-x 2 root root 4096 Dec 4 15:24 VoiceMail Office ./INBOX: total 0 ./Old: total 0 ./temp: total 0 ./tmp: total 0 ./VoiceMail Office: total 0 but from the CLI I get :- voip*CLI voicemail show users ContextMbox User Zone NewMsg voicemail 1001 user2 Best Regards, - --[ UxBoD ]-- ux...@splatnix.net wrote: | [r...@voip ~]# asterisk -V | Asterisk 1.6.1.11 | | When using the above version with IMAP VoiceMail integration when I | leave a message my SNOM360 it shows 2 message waiting; yet when | running voicemail show users from the Asterisk CLI it correctly | reports 1. | | It would appear that when the VM is temporarily stored, and the VM is | delivered by IMAP to the remote mail account, the MWI is being | initiated with a incorrect count. | | I then delete the VM from either 1) the phone 2) the mail account the | MWI goes blank and the message count shows 0 correctly. | | I am still trying to debug but any thoughts on this ? | | Here is how I have voicemail.conf :- | | [general] | format=wav49 | maxsecs=180 | minsecs=5 | skipms=3000 | maxsilence=3 | silencethreshold=128 | maxlogins=3 | imapserver=imap_server | imapfolder=VoiceMail Office | imapport=993 | imapflags=ssl | authuser=imap_user | authpassword=imap_password | | [voicemail] | 1001 = 1234,user,,,imapuser=u...@imap_server | | Best Regards, | | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI count wrong when using IMAP and VM
as soon as I delete the two messages I receive in the console :- [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning: Unknown message data: 1 EXPUNGE [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning: Unknown message data: 1 EXPUNGE Best Regards, - --[ UxBoD ]-- ux...@splatnix.net wrote: | [r...@voip ~]# asterisk -V | Asterisk 1.6.1.11 | | When using the above version with IMAP VoiceMail integration when I | leave a message my SNOM360 it shows 2 message waiting; yet when | running voicemail show users from the Asterisk CLI it correctly | reports 1. | | It would appear that when the VM is temporarily stored, and the VM is | delivered by IMAP to the remote mail account, the MWI is being | initiated with a incorrect count. | | I then delete the VM from either 1) the phone 2) the mail account the | MWI goes blank and the message count shows 0 correctly. | | I am still trying to debug but any thoughts on this ? | | Here is how I have voicemail.conf :- | | [general] | format=wav49 | maxsecs=180 | minsecs=5 | skipms=3000 | maxsilence=3 | silencethreshold=128 | maxlogins=3 | imapserver=imap_server | imapfolder=VoiceMail Office | imapport=993 | imapflags=ssl | authuser=imap_user | authpassword=imap_password | | [voicemail] | 1001 = 1234,user,,,imapuser=u...@imap_server | | Best Regards, | | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Sunday 18 October 2009 20:04:40 John A. Sullivan III wrote: On Sun, 2009-10-18 at 19:14 -0500, Tilghman Lesher wrote: On Thursday 15 October 2009 20:13:55 John A. Sullivan III wrote: On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John No, comma is the right delimiter, unless you're using ODBC storage for voicemail, in which case, I'm terribly sorry, but multiple mailboxes are not supported in that line. This has been corrected in SVN for all 1.6 branches. I'm not using ODBC but I am using IMAP. Could that be the problem? No, the IMAP code supports multiple mailboxes fine. Clearly, you have another problem that has yet to be diagnosed correctly. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Thursday 15 October 2009 20:13:55 John A. Sullivan III wrote: On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John No, comma is the right delimiter, unless you're using ODBC storage for voicemail, in which case, I'm terribly sorry, but multiple mailboxes are not supported in that line. This has been corrected in SVN for all 1.6 branches. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Sun, 2009-10-18 at 19:14 -0500, Tilghman Lesher wrote: On Thursday 15 October 2009 20:13:55 John A. Sullivan III wrote: On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John No, comma is the right delimiter, unless you're using ODBC storage for voicemail, in which case, I'm terribly sorry, but multiple mailboxes are not supported in that line. This has been corrected in SVN for all 1.6 branches. I'm not using ODBC but I am using IMAP. Could that be the problem? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
No, probably my ignorance but why would I do that? I set up all the users, extensions, and mailboxes manually by editing the config files in order to have more control than the user.conf gives me (if I understand the user.conf file properly - I've never used it based upon reading the documentation). Thanks - John On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote: I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Preparing for the lightning bolt here (ready to duck!!), the way I have things set up, tkeeley would have an entry in users.conf as 612 and 610 would have an entry in users.conf as 610. There would be an entry in sip.conf for tkeeley under 612 and no entry for 610 since it's just a mailbox and not a physical extension. Not necessarily best or even correct, just works for me. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes No, probably my ignorance but why would I do that? I set up all the users, extensions, and mailboxes manually by editing the config files in order to have more control than the user.conf gives me (if I understand the user.conf file properly - I've never used it based upon reading the documentation). Thanks - John On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote: I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser= morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
The secondary value is used, just not by the MWI functionality. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
No, I'm saying you need two tkeeley entries with one mailbox each. The multiple entry is fine for other mailbox functionality, just not MWI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI for multiple voice mail boxes
Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI
I have always been confused about how MWI is working with asterisk. If a SIP device has the option to subscribe to MWI, should it? I think asterisk sends NOTIFY messages to SIP clients if the sip peer entry has mailbox=. Is there any advantage then to leaving that out of the sip peer entry and having the device itself register for the messages? Is it really one and the same thing? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI
Jeff LaCoursiere wrote: I have always been confused about how MWI is working with asterisk. If a SIP device has the option to subscribe to MWI, should it? I think asterisk sends NOTIFY messages to SIP clients if the sip peer entry has mailbox=. Is there any advantage then to leaving that out of the sip peer entry and having the device itself register for the messages? Is it really one and the same thing? It's really one and the same thing. NOTIFY messages can only be sent to UAs that have SUBSCRIBE'd to them. Asterisk only accepts subscriptions to the MWI events if there is a mailbox entry in the SIP peer. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI using Asterisk and external mail server
We are upgrading to Asterisk as our company phone switch. We have our own Auto-Attendant, call center and voice mail servers. One issue that we are having right now is our voice mail server sending Asterisk the NOTIFY to turn MWI on our Polycom SIP phones. I see from the research I've done (including previous posts here) that Asterisk did not support external MWI events. Is this still the case? Is there anything in the works currently to change that? In the end, all we are looking for is to have the ability to turn on and off the MWI light on the SIP phones. Thanks, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI Asterisk+Openser
Hi, I need some help, getting to work asterisk MWI. I set up Asterisk as voicemail server for Openser as this tutorial shows : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I debug it? Thank you Szasz Szabolcs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
2008/10/28 Robert Boardman [EMAIL PROTECTED] Olivier wrote: 2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? Regards Replying to myself, for an unknown reason, MWI is weirdly working : - Phone icon inconsistently shows awaiting voicemails, - NOTIFY message from Asterisk are still replied with 481 Call Leg/Transaction Does Not Exist When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail Notifications : - you can see SUBSCRIBE message - you can see NOTIFY answer - you can't see any 481 Call Leg/Transaction Does Not Exist reply to this NOTIFY message From then on, further NOTIFY messages are replied with 481 Call Leg/Transaction Does Not Exist and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db To: sip:sip:[EMAIL PROTECTED]:5060 http://sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/0 (0/0) NOTIFY message rejected by S450IP (rejected means 481 reply) NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport From: asterisk sip:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=as5e574490 To: sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 96 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Voice-Message: 3/0 (0/0) The only difference I see between both is that new NOTIFY don't include : Subscription-State: active Do you see something else ? Is it possible to easily add this Subscription-State field without patching Asterisk source (I'm unable to do that) ? Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just worked out a good way of getting transfer working Using features .conf [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer DTMF A-D are valid DTMF signals but are not usually shown on standard phones so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb Hi, What about MWI and Subscription-State: active ? I can see that Asterisk sends NOTIFY messages with and without this Subscription-State: active statement in header. I can see that NOTIFY messages without Subscription-State: active are rejected by Gigaset base station. Is it possible to either configure : 1. Gigaset to accept NOTIFY messages without Subscription-State: active 2. Asterisk to send NOTIFY messages with Subscription-State: active Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
Olivier wrote: 2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? Regards Replying to myself, for an unknown reason, MWI is weirdly working : - Phone icon inconsistently shows awaiting voicemails, - NOTIFY message from Asterisk are still replied with 481 Call Leg/Transaction Does Not Exist When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail Notifications : - you can see SUBSCRIBE message - you can see NOTIFY answer - you can't see any 481 Call Leg/Transaction Does Not Exist reply to this NOTIFY message From then on, further NOTIFY messages are replied with 481 Call Leg/Transaction Does Not Exist and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db To: sip:sip:[EMAIL PROTECTED]:5060 http://sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520 Contact: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/0 (0/0) NOTIFY message rejected by S450IP (rejected means 481 reply) NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport From: asterisk sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED];tag=as5e574490 To: sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 96 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Voice-Message: 3/0 (0/0) The only difference I see between both is that new NOTIFY don't include : Subscription-State: active Do you see something else ? Is it possible to easily add this Subscription-State field without patching Asterisk source (I'm unable to do that) ? Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just worked out a good way of getting transfer working Using features .conf [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer DTMF A-D are valid DTMF signals but are not usually shown on standard phones so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
2008/10/3 Olivier [EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? Regards Replying to myself, for an unknown reason, MWI is weirdly working : - Phone icon inconsistently shows awaiting voicemails, - NOTIFY message from Asterisk are still replied with 481 Call Leg/Transaction Does Not Exist When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail Notifications : - you can see SUBSCRIBE message - you can see NOTIFY answer - you can't see any 481 Call Leg/Transaction Does Not Exist reply to this NOTIFY message From then on, further NOTIFY messages are replied with 481 Call Leg/Transaction Does Not Exist and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db To: sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/0 (0/0) NOTIFY message rejected by S450IP (rejected means 481 reply) NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport From: asterisk sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=as5e574490 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 96 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED][EMAIL PROTECTED] Voice-Message: 3/0 (0/0) The only difference I see between both is that new NOTIFY don't include : Subscription-State: active Do you see something else ? Is it possible to easily add this Subscription-State field without patching Asterisk source (I'm unable to do that) ? Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
Kevin P. Fleming wrote: Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. do you have a suggestion, there is only two fields that can be filled in that to refer to the R key, Application-type: I think this is content type Application-signal: what it sends? Thanks for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI with Siemens Gigaset S450IP
Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
Karl Fife wrote: Do I understand correctly that we are not talking about redundant MWI status traffic here, we're ONLY talking about the notion that asterisk ignores MWI subscription status and behaves as if it has 100% MWI subscription. That is unless subscribemwi=yes is in sip.conf. Is that an accurate summary? Yes. And theoretically, if I had thousands of endpoints and I needed 100% mwi subscription, there may be some theoretical efficiency to turning off all MWI subscriptions in all of the endpoints. Likewise if only 25% of my 'thousands' of endpoints needed any MWI, there would be some efficiency in setting subscribemwi=yes, and explicitly subscribing only those 25%. Is that right? Theoretically, I guess so. However, keep in mind that this is really about dealing with odd specifics of how certain phones behave more than anything else. Some phones won't subscribe but still expect MWI. Some phones may freak out if they receive MWI when they haven't subscribed to it. Thanks again for clarifying! I appreciate it! You are quite welcome. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI working perfectly. Shouldn't it be broken??
I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but Sip show subscriptions does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM Polycom endpoints. I'm confused. Isn't MWI subscription the method by which the device is setting its MWI? I know that Asterisk 1.6 is moving toward an event-driven model, but I'm running 1.4.21.1, so why is this working? I know that some smart cookie on this list will know the answer, but unfortunately I am not said cookie. FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
On Aug 24, 2008, at 11:29 AM, Karl Fife wrote: FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Asterisk will send the NOTIFY for MWI even if the device doesn't subscribe, unless you tell it not to. This is necessary for some phones for MWI to work. If you _don't_ want Asterisk to do this, you can set the subscribemwi=yes option in sip.conf. This tells Asterisk to _only_ send MWI with an associated subscription. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
I see. Thanks Russel. And I now notice that if I explicitly tell my 962 the IP address of the mail server, it will also subscribe. Do I understand correctly that we are not talking about redundant MWI status traffic here, we're ONLY talking about the notion that asterisk ignores MWI subscription status and behaves as if it has 100% MWI subscription. That is unless subscribemwi=yes is in sip.conf. Is that an accurate summary? And theoretically, if I had thousands of endpoints and I needed 100% mwi subscription, there may be some theoretical efficiency to turning off all MWI subscriptions in all of the endpoints. Likewise if only 25% of my 'thousands' of endpoints needed any MWI, there would be some efficiency in setting subscribemwi=yes, and explicitly subscribing only those 25%. Is that right? Thanks again for clarifying! I appreciate it! -Karl On Sun, 24 Aug 2008 11:50:19 -0500, Russell Bryant [EMAIL PROTECTED] said: On Aug 24, 2008, at 11:29 AM, Karl Fife wrote: FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Asterisk will send the NOTIFY for MWI even if the device doesn't subscribe, unless you tell it not to. This is necessary for some phones for MWI to work. If you _don't_ want Asterisk to do this, you can set the subscribemwi=yes option in sip.conf. This tells Asterisk to _only_ send MWI with an associated subscription. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI for voicemail - H323
Hi , How does the Asterisk provide Voicemail Message waiting indication to an h323 endpoint configured with Asterisk. Please provide the required Setup / comfiguration details or redirect to appropriate to resource. Awaiting an earliest positive response. Thanks in advance, Anisha ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
The 481 Call Leg/Transaction Does Not Exist response to the NOTIFY makes me think that you might need to configure the phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages from the phone when it is booted? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaap Winius Sent: 13 February 2008 18:46 Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Quoting Steve Langstaff [EMAIL PROTECTED]: The 481 Call Leg/Transaction Does Not Exist response to the NOTIFY makes me think that you might need to configure the phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages from the phone when it is booted? Yeah, sure. And there are some error messages mixed in too: == 14:01:23.425955 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 473 ... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.426075 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 509 [EMAIL PROTECTED] ...vSIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:23.480238 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..k... ..F.SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.480375 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 14:01:23.918830 arp who-has gigaset.umrk.to tell bitis.umrk.to ../.E .. 14:01:23.921726 arp reply gigaset.umrk.to is-at 00:01:e3:77:f8:67 (oui Unknown) ...w.g../.E .. 14:01:24.539636 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 476 E.. ..2gSUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.539816 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 512 [EMAIL PROTECTED] ...ySIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:24.594442 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..i... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.594557 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 E...- [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 == Before this was a series of REGISTER messages, and afterwards a series of OPTIONS messages. However, no errors there. Also, this is without having set 'mailbox=1000' or '[EMAIL PROTECTED]' in /etc/asterisk/sip.conf. And, now that I look at it again, the network mailbox settings for the Siemens phone won't have anything to do with these errors either, since it simply makes it possible to associate a button on each handset with an extension used to access a voicemail account. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI problem with Siemens Gigaset S675 IP
Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is) I've had to add the voicemail context to get MWI to work correctly in the past. - Original Message - From: Jaap Winius [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:45 PM Subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Quoting Henry Devito [EMAIL PROTECTED]: Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is) I've had to add the voicemail context to get MWI to work correctly in the past. According to the documentation, you shouldn't have to add @context if the context is 'default'. But, I went ahead and tried it out anyway. I even tried using some other context names, but it makes no difference: the error remains the same. Thanks anyway, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mwi with sip
Hi, I am trying to utilize MWI with sip channel. when my client sens a SUBSCRIBE to asterisk I get info that user not found: - [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip (domain ms.sip.rd.touk.pl) [Jan 28 11:49:02] --- Transmitting (no NAT) to 192.168.129.38:7060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.129.38:7060;branch=z9hG4bKadcf.1752dae4.0;received= 192.168.129.38 Via: SIP/2.0/UDP 192.168.0.165:7360;rport=7360;branch=z9hG4bKdxcekurc From: hellboy sip:[EMAIL PROTECTED];tag=qrrlr To: hellboy sip:[EMAIL PROTECTED];tag=as70810877 Call-ID: [EMAIL PROTECTED] CSeq: 968 SUBSCRIBE User-Agent: TouK S.K.A Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 does user have to be registered in asterisk I am using asterisk as media server but my users are registered at other sip proxy. Please point me what do I miss? Best regards tomasz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, My problem was that the context wasn't the same in my voicemail.conf and in my sip.conf!! One was 'default' and the other 'device' I have put 'default' everywhere and it's working! Have a nice day Jared Smith a écrit : On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote: It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... Do you have a mailbox defined for the SIP device in sip.conf? If you don't, Asterisk has no way of matching up a mailbox to a particular SIP device. -Jared Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVmcvN4+o+2LtdFwRAkdSAJ9KPkr9NGc9nm+wIFGUofcE4nxQnACfRJeL HakgTsDpHM7QCCyvzPI0440= =J5cK -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... my voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; 6710 = 1234,Compte Test 0,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes|saycid=yes|envelope=yes|delete=no Alex Balashov a écrit : Sorry, not sure I understand the question. What is the problem here? On Mon, 3 Dec 2007, Marc LEURENT wrote: Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 - 192.168.95.73:5060 NOTIFY sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK114bbd0e;rport. f: Unknown sip:[EMAIL PROTECTED];tag=as5087d720. t: sip:[EMAIL PROTECTED]:5060;user=phone. m: sip:[EMAIL PROTECTED]. i: [EMAIL PROTECTED] CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. o: message-summary. c: application/simple-message-summary. l: 94. . Messages-Waiting: no. Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0). ___ - --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVX5jN4+o+2LtdFwRAicZAKCwjAojZxq6gbF2+qvyUozYteBwMACfZq51 WqddUJCEAI7Q18V3ROv0FVk= =tKYm -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI error
On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote: It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... Do you have a mailbox defined for the SIP device in sip.conf? If you don't, Asterisk has no way of matching up a mailbox to a particular SIP device. -Jared Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 - 192.168.95.73:5060 NOTIFY sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK114bbd0e;rport. f: Unknown sip:[EMAIL PROTECTED];tag=as5087d720. t: sip:[EMAIL PROTECTED]:5060;user=phone. m: sip:[EMAIL PROTECTED]. i: [EMAIL PROTECTED] CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. o: message-summary. c: application/simple-message-summary. l: 94. . Messages-Waiting: no. Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0). -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVEeaN4+o+2LtdFwRAhfGAJ4/iL4yG0xm5XBaYLUxGzpgKitGNwCfREV+ H9wJ6bD+ITOBDoKm2gstEQQ= =3MmR -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI error
Sorry, not sure I understand the question. What is the problem here? On Mon, 3 Dec 2007, Marc LEURENT wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 - 192.168.95.73:5060 NOTIFY sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK114bbd0e;rport. f: Unknown sip:[EMAIL PROTECTED];tag=as5087d720. t: sip:[EMAIL PROTECTED]:5060;user=phone. m: sip:[EMAIL PROTECTED]. i: [EMAIL PROTECTED] CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. o: message-summary. c: application/simple-message-summary. l: 94. . Messages-Waiting: no. Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0). -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVEeaN4+o+2LtdFwRAhfGAJ4/iL4yG0xm5XBaYLUxGzpgKitGNwCfREV+ H9wJ6bD+ITOBDoKm2gstEQQ= =3MmR -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI across multiple servers
Jon, I would be as well very interested in your Voicemail Solution : AGI + Web Interface to retrieve voice messages. By the way, you sotre to MySQL, do you use ODBC for that ? or something else, in that case, what ;o) ? Thanks in advance ! Jean-Marc On 12/7/06, Jon Farmer [EMAIL PROTECTED] wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
On 8 Dec 2006, at 15:02, Jean-Marc Salsa wrote: Jon, I would be as well very interested in your Voicemail Solution : AGI + Web Interface to retrieve voice messages. By the way, you sotre to MySQL, do you use ODBC for that ? or something else, in that case, what ;o) ? Thanks in advance ! Jean-Marc On 12/7/06, Jon Farmer [EMAIL PROTECTED] wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards You might want to look at integrating my (free opensource) gsmPlay applet into the web front end of that, it would let your users play their gsm voicemails without installing quicktime... Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
On Dec 7, 2006, at 4:14 AM, Jon Farmer wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon, Maybe you could post this application and a how-to to the wiki? Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI across multiple servers
Jon: I will second that motion ... This is something I would be very interested in seeing as I have a similar requirement ... Have a number of folks on my system who work from home ... A number of them have Asterisk servers that register with the main office Asterisk server ... Right now I am handing off calls to each Asterisk server so the VM gets recorded locally and I can make the MWI light blink ... Only reason I did it that wasy is because I was not smart enough to figure out how to centralize VM at the main office and still turn on that darn MWI blinky light ... Are you able in your scenario to store all VM on a central server, but some how get the word back to the remote server that there is a message waiting ??? If so that is col !!! I spent days trying to figure out how to do that and finally just gave up ... PLEASE POST THAT ONE ON THE WIKI If you don't have time to write up a how-to, at least post your scripts with a quick and dirty of what it does ... Maybe make it searchable by remote MWI or something similar ... G.Hendershot ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI/realtime/openSer in 1.4
Hi, can somebody clear up the situation with SIP voicemail SUBSCRIBE and realtime SIP peers in 1.4 for me. From a lot of sketchy information about the new chan_sip in 1.4 I know that it implements rfc compliant SUBSCRIBE behaviour (with the subscribemwi=yes option?), but what about realtime peers? Since realtime peers are only created once the peer registers, will a UA that SUBSCRIBEs to voicemail receive NOTIFYs even if it is a realtime peer that is REGISTERed with openSER in front of asterisk (and therefore does not exist a a peer in asterisk)? Can someone point me to some good (or any) docs on the subscribe implementation with realtime in 1.4? tx M PS: I know of the possible workarounds with sipsak/scripts/etc but that does not work with all phones and SIP stacks and I would like to get it working with SUBSCRIBE on 1.4 if possible. -- Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI across multiple servers
We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
We've got a setup similar to that. Depending on how you want to set it up, just use externnotify and a script that touches the msgnum.txt file in the user's vm directory on the other boxes. We're using ssh, you may choose to use a different method. It's an immediate MWI notification, and seems to work well. If you're interested, let me know, I'll shoot the scripts over to you. On Wed, 2006-12-06 at 09:20 -0700, Porier, Jeremy M. wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MWI across multiple servers
Aaron, Yeah, could you please send me that script. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, December 06, 2006 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers We've got a setup similar to that. Depending on how you want to set it up, just use externnotify and a script that touches the msgnum.txt file in the user's vm directory on the other boxes. We're using ssh, you may choose to use a different method. It's an immediate MWI notification, and seems to work well. If you're interested, let me know, I'll shoot the scripts over to you. On Wed, 2006-12-06 at 09:20 -0700, Porier, Jeremy M. wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
Aaron, Could you please send me the scripts as well. Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
I've posted the instructions and scripts on my blog for everyone to grab. This way I'm not sending random files to random people :) http://asterisk.mdaniel.net/?p=14 Let me know if I need to change anything. On Wed, 2006-12-06 at 10:12 -0700, David Thomas wrote: Aaron, Could you please send me the scripts as well. Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MWI across multiple servers
Been working fine for us so far. -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
Got it mate. thanx for that. Am using mysql for voicemail storage, unlike in the script you've written which works on mails on disk on a certain path. All I've to do is query for INBOX(new) and Old(old) voicemessages count. cheerz - Ben Scott Keagy wrote: A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mwi for voicemail not showing up for realtime config.
Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363highlight= Regards, Scott -Original Message- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
