[asterisk-users] Asterisk 11 with Dovecot IMAP Email
Hi all, Has anyone gotten dovecot 2.2 working with the asterisk 11 imap voicemail system? I can login via thunderbird or telnet to the imap server just fine. The below is logged: [Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:3176 mm_log: IMAP Error: Can't open mailbox {localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX: invalid remote specification [Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2906 init_mailstream: Can't connect to imap server {localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX [Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2424 __messagecount: Houston we have a problem - IMAP mailstream is NULL As of now its not even sending a imap request to dovecot. I am using a master user which I can successfully login as. Ive tried specifying individual usernames/passwords in the voicemail.conf file. This doesnt work either. Thanks --Tammy signature.asc Description: OpenPGP digital signature -- _ -- 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] Asterisk & Vitelity Invite issues
my bad, both sides are generating re-invites. Vitelity ignores any inbound invites to continue call flow. to keep the call going our pbx has to deal with their re-invites otherwise the call terminates at 30 minutes on the dot. Our side is ignoring the inbound invites from vitelity and that causes the call to be torn down. On 8/10/16 4:21 PM, Matt Fredrickson wrote: > Wait a second, I thought in your original email that you said that > Asterisk was generating reinvites. It sounds now like you're saying > that the remote side is initiating reinvites instead. > > My understanding is that the canreinvite/directmedia option only > influences Asterisk's behavior with regards to generating reinivites. > If it receives a reinvite, I don't think these options will do > anything about that. In fact, I'd guess that not properly responding > to a received reinvite is going to potentially break things from the > SIP perspective. > > Matthew Fredrickson > > > On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly <tammy-li...@wiztech.biz> > wrote: >> >> >> On 8/9/16 12:40 PM, Matt Fredrickson wrote: >>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> >>> wrote: >>>> Hi All, >>>> >>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >>>> split off to where they need to go. We are having a problem getting >>>> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up >>>> sending a reinvite which their side & they do not support us sending a >>>> reinvite. Ive tried: >>>> >>>> canreinvite=no which was supposedly replaced by: >>>> >>>> directmedia=no >>>> >>>> Can anyone shed any light on this matter? I'd love to get this fixed. >>>> >>> >>> Those options *should* influence chan_sip's reinvite behavior - at >>> least they have from my experiences working with chan_sip. Do you >>> know what is triggering the reinvite in the first place, or does it >>> look like a normal media reinvite? >>> >> >> >> every 15 minutes vitelity sends a re-invite to keep the call going. I >> have a packet capture from it if you'd like it feel free to email me off >> list @ tamara.wis...@wiztech.biz >> >> -- >> _ >> -- 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up >> sending a reinvite which their side & they do not support us sending a >> reinvite. Ive tried: >> >> canreinvite=no which was supposedly replaced by: >> >> directmedia=no >> >> Can anyone shed any light on this matter? I'd love to get this fixed. >> > > Those options *should* influence chan_sip's reinvite behavior - at > least they have from my experiences working with chan_sip. Do you > know what is triggering the reinvite in the first place, or does it > look like a normal media reinvite? > every 15 minutes vitelity sends a re-invite to keep the call going. I have a packet capture from it if you'd like it feel free to email me off list @ tamara.wis...@wiztech.biz -- _ -- 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] Asterisk & Vitelity Invite issues
Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed any light on this matter? I'd love to get this fixed. There is no firewall on this machine at all. Thanks --Tammy -- _ -- 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] Please check Important..
This is a phishing URL. Dont click it. Moderators: can you put this lovely person on the moderated list? On 9/25/13 21:11:44, Code Lover wrote: Hi, I have attached an important document via Google docs, Please check the link below for additional security you will be required to sign in with your email before viewing / downloading the document. *removed* -- Thank You, Abdul Lateef Senior – Development Barwa Bank Doha Qatar --- Please do not print this e-mail unless it is absolutely necessary. -- _ -- 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