similar to: Asterisk not sending bye message to original UA

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

On 06.09.2018 22:25, kadafax at gmail.com wrote: > I necro-bump this thread as I have the same problem since I switched to LMTP from LDA (as the wiki recommend). > Any news to make dovecot LMTP postix compliant ? > > > Le 08/11/2016 ? 17:13, Noah Tilton a ?crit?: >> >> I was wondering whether there is a roadmap for adding SMTPUTF8 support to Dovecot? >> >>

I necro-bump this thread as I have the same problem since I switched to LMTP from LDA (as the wiki recommend). Any news to make dovecot LMTP postix compliant ? Le 08/11/2016 ? 17:13, Noah Tilton a ?crit?: > > I was wondering whether there is a roadmap for adding SMTPUTF8 support > to Dovecot? > > My delivery pattern is Postfix -> Dovecot LMTP and it is choking on > utf8

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

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The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

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I came across a quite interesting post from Ross Ihaka, thought would be good to share it and get the opinion of folks around here. I am not sure where to post this for the R community but since it has to do with development I thought or R-devel Ross Ihaka Newsgroups: comp.lang.lisp From: Ross Ihaka <ih... at stat.auckland.ac.nz> Date: Wed, 23 Jan 2008 10:35:26 +1300 Local: Tues, Jan 22

Hi, Let me explain what I'm looking for a solution using asterisk. I have one third party SIP based server (A) and on Asterisk server (B). 1. Extension-1 --> Server A calls Server B. 2. Server B does some processing and calls/sends back to Server A ---> Extension-2 3. SIP session has been established b/w two Extension-1 and Extension-2. Now is there any config that I can do in

Good evening list, I'm looking for an R implementation of the "Shuffled Complex Evolution??? (SCE-UA) algorithm after Duan et al. (1993). Does anybody know if there is an extension/ package existing that contains it? Thanks very much for your help! Cheers, Simon Duan QY, Gupta KV, Sorooshian S (1993) Shuffled Complex Evolution Approach for Effective and Efficient Global Minimization. In

Hi, I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine. I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is

I've got the following situation where a UA is trying to call another UA via Asterisk and SER according to UA1 -> * -> SER -> UA2. Now in the event that SER generates a 404 Not Found for UA2 I would like Asterisk to return or relay or forward or whatever the 404 to UA1. Anyone know this might be able to be done (or maybe not possible at all?) Craig

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

Hi I registered my asterisk box to my SIP provider as an UA. For every call I receive on this trunk, I get the message "That is not a valid conference number". I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. Szasz Szabolcs -------------- next part

Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ?externaddr?, ?localnet? and ?nat=force_rport,comedia?. Asterisk registration is successful, I see in Wireshark the packets send between Asterisk and SIP server. However,

Morning! I'm trying to subscribe a softphone to an Asterisk 11 server, but it sends an "ua-profile" event that Asterisk immediately rejects with a 489 Bad Event error. Is this event not supported at all? Are there any workarounds? Best regards, Norman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! As reported earlier this week, I have problems with a sometimes-crashing Asterisk. In most of the cases safe_asterisk is able to restart it. But sometimes it crashes, so that manual interaction is necessary. The seg-faults and crashes occurs, right after call between a SIP Terminal and a legacy PSTN Terminal (PRI/Euro-ISDN), but only if the SIP Terminal hangs up as first. No problem, if the

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I