similar to: Asterisk and G.726 Codec

Hello all, For some years now, I've been using Thunderbird for dovecot. I am not very satisfied with t/b so I thought of using m/s outlook but then I thought that I want to distance my clients from office products. I have a newly created dovecot installation on a very small site. Three nodes, all x86 Windows 7 professional with an ubuntu v14.04 server (x86 again) running dovecot 1.2.17.

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

Dear All, i have following CLI error while try to run this command from Dialplan *TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute 'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into

Boa Tarde Lista. I'm having problems in tramiss?o a fax using T.38. My scenario is: Asterisk 1.6.0.5 2 ATA of Intelbras 2210. ReceiveFAX in the asterisk. Unable to fax when it is a ATA to another user on the Asterisk means, if I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX of Asterisk works perfect, but if I try to pass between two Branches

Hi, I have both codec_g726.so and format_g726.so loaded: root at test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0

I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones were dropped after a minute or so. I installed 1.6.1-rc1 and now the newer Aastra phones (5xi) work properly. The problem remains with the older phones (9112i, 9133i and 480i). If I dial

Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

Hi, I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the cli, I look at the channel variables and I can see the new status, but que it hang-ups the CDR doesn't have this value. I'm using mysql backend for cdr Any idea? Thnks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

hi all, I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed when i leave voicemail On Asterisk i need MWI Indication on kamailio extension there are some methods i tried but still cant get success All other feature are working fine also try voip-info.org methods can anybody suggest me for different method and have some different setting on SIP . any help appreciated

hello, I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read application accepts # sign, So is it possible? And maybe there is a workaround? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/2f2b443c/attachment.htm

Hi, I am trying to use asterisk open source version(asterisk-1.6.0.5) with MySQL (using res_odbc)support for extensions and users list. The call rate is 7 calls / second and each call stays for 120 seconds. after making 25000 successful calls , calls started failing with following message on CLI. [Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable to create channel of

hello All, i have one issue with Asterisk Meetme Application i am recording through Meetme channels through option *'r'* and format for recording a file is '*wav*' lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. i have very strange problem of meetme_recording , once conference starts recording file having a *recording is 2x faster *than normal recording .

Hi, I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf configured and modules.conf have preload => res_odbc.so preload => res_config_odbc.so extconfig.conf has queue_log => odbc,asterisk. When I start asterisk I get the following messages. The important one being: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available

Hi all! I would like to make a service with recording sounds and playing back to caller. I had wrote the script but it failed at Read statement with file not found error. I have put some file test into script and this is what happen on verbose level 9. -- Executing [8298 at default:8] Record("DAHDI/27-1",

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hi everybody, In an Asterisk 1.6.0.5, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, with a Digium B410P and an OpenVox A400E, I can't make "dahdi_scan" to recognize the OpenVox. This card was working correctly but suddenly stopped working and I cannot make it work again. Both ?lspci? and ?dahdi_hardware? detect it but ?dahdi_scan? not and I cannot use it. >lsppci: *0a:00.0

I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be