similar to: Error - Failed to extend from xxx to xxx

Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

Hi all, I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server using freetds and unixodbc, which works with 1.6.1.20. With the same config in 1.8 I get an error when trying to start asterisk which says: [Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined

Hello, In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were committed between versions 1.6.1.1 and 1.6.1.2. But if I'm not mistaken, you cannot read anything there about Asterisk to Asterisk-addons compatibility. What is the rule for Asterisk to Asterisk-addons compatibility ? Is this rule implicit ("any Asterisk-addons 1.6.1.X is compatible with any Asterisk

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

I'm using a Asterisk 1.6.1.6 with dahdi. We need to redirect phone calls to a certain number when there is nobody. So I read about call reflection but the call reflection applications on bristuff are not for 1.6.1.6. Are there any other applications or patches that provides call reflection for Asterisk 1.6.1.6?? Greetz TM

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Hello,

Hello, When I create a sip peer? in users.conf then a hint is automatically created for that peer. But when I create a peer in sip.conf or a realtime peer with the same values then this hint is not created. Every time I add such peers I have to add a hint in extensions.conf. Is it possible to have something like?? exten => _XXX,hint,SIP/${EXTEN}? in extensions.conf so that I don't have

It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 301 Moved

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

Hi all, Anyone know why this happens? Mem: 524288k total, 508120k used, 16168k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hello everyone! This list may not be entirely appropriate discussion, but in the lack of ogg@xiph.org or ogg-dev@xiph.org this will have to do. I've been thinking for a few weeks that Ogg needs a simple text stream (read subtitle) format to go along with theora. This is important, because otherwise I can't transcode fellowship of the rings while keeping the elvish-speek, unless I render

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When