similar to: Asterisk removes a charachter from sip peer name

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olli Heiskanen Sent: 03 January 2015 08:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk removes a charachter from sip peer name Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names

Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hello, I'm stuck with getting cdr records stored in MySql database. I have a working realtime environment and have verified that the db connection works fine when used via res_config_mysql.conf. I'd appreciate Your help on how to get the odbc connector working as I think there's something wrong with its configuration. The problem presented itself as an error when making a call that

Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

Is there any way to put a newline character in comments for ogg vorbis files in Linux? I can't see any way of doing it with vorbiscomment. Easytag would work, but has other problems (like concatenating comment fields with the same tag name). If I wanted to put together a quick hack to add a such a comment (I'm thinking read the comment from a file, with the tag name specified at the

Is there any way to put a newline character in comments for ogg vorbis files in Linux? I can't see any way of doing it with vorbiscomment. Easytag would work, but has other problems (like concatenating comment fields with the same tag name). If I wanted to put together a quick hack to add a such a comment (I'm thinking read the comment from a file, with the tag name specified at the

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using

On 6/5/2020 12:21 PM, Johnny Hughes wrote: > if you click on the six digit number, for example, e52775 for the > current latest "import 389-ds-base-1.3.10.1-9.el7_8". The result is > every diff of every change for the rpm. That's quite handy! But not what I'm looking for. I'm trying to figure out what edits I made to my config files. My most recent case was

On Fri, Aug 30, 2019 at 12:17:47PM +0100, Gary Stainburn wrote: > On Friday 30 August 2019 12:03:26 Alexander Dalloz wrote: > > > > Besides a corrupted certificates bundle I cannot imagine a different > > root cause actually. Just to mention that the 'etckeeper' package from EPEL is great for tracking changes to /etc. Package installs trigger a commit, as do a daily

Hi, I've been trying to get xen to work on a couple of old machines for a while now. it runs for a week or two, and then when it stops the machine powers off and I find I have to hold the front power switch in for ~10 secs before toggling the back power switch makes any difference. clues ? Regards, Paddy

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

I'm writing a script to sync some mp3 files. Due to a limitation in the number of destination files that can be read from my thumb drive, I'm not looking to preserve the original file structure (actually, I'm looking to sync *only the files* to the new destination directory). The source files are all subfolders under /backup/Music: ./Adrian Legg/Mrs. Crowe's Blue Waltz/Paddy

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to