[asterisk-users] (no subject)

to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Dealing with muti-body INVITE

INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Asterisk crash

Is the problem reproducable? /Adnan On Tue, Sep 3, 2013 at 11:17 AM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, ** ** In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home

[asterisk-users] Not able to post to list

Hi who is responsible for this mailing list? i am not able to post to it. Br Adnan Sent from my iPhone On 9 okt 2012, at 21:04, Matthew Jordan mjor...@digium.com wrote: On 10/09/2012 02:00 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of libpri

Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers

the solution lies in kamailio/opensips's despatcher module. Sent from my iPhone On 23 maj 2012, at 20:46, bilal ghayyad bilmar...@yahoo.com wrote: Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000

Re: [asterisk-users] Asterisk log format

grep or sgrep Sent from my iPhone On 15 dec 2011, at 18:46, Asterisk Guy arpexpe...@gmail.com wrote: Hi mates! Please, I need to understand how to search for an specific log by date/time on asterisk logs, but can't understand how this works, can you guys please give me an example about

[Asterisk-Users] asterisk1.2.1/PRI-E1 outbound call issues

Hello List,I have a following setup:1-Intel Zeon 3.0 Ghz dual Zeon capable board2-Ram 1GB3-OS SLES9 SP24-Asterisk-1.2.15-Wildcard TE110P(Using as E1) 6-Wildcard TDM03BPRI/E1 is up and running perfectly inbound /outbound calls goes perfectly in start but after sometime almost all outbound calls

Re: [Asterisk-Users] Video Conferencing

On 1/1/06, Nir Simionovich [EMAIL PROTECTED] wrote: Well, the documentation states that Video Conferencing is possible. I'vetried working with EyeBeam, which yielded nice Results, but anything beyondthat - I can't comment.Nir Scan you share your experience with us i.e. what asterisk version what

Re: [Asterisk-Users] Very complicated dialplans?

On 8/6/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Peter Svensson wrote: On Sat, 6 Aug 2005, Robert Goodyear wrote:Can you educate us all on the appropriate circumstances in which touse 'r'? Some devices (voip phones, softphones) do not generate in band progress information when

[Asterisk-Users] Asterisk wiht LDAP

I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the

[Asterisk-Users] Asterisk wiht LDAP

I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the

[Asterisk-Users] Single extension/user registers across multiple asterisk servers

Hello , I have a question which i am not clear that whether it is possible or not so i want some help to clearify Sorry for very long mail: we have eight asterisk servers across different cities connected through IAX intenet connection is DSL broadband so for sake simplicity and easiness for eight

Re: [Asterisk-Users] CRM integration (was RE: CallerID)

On 5/31/05, Anton Krall [EMAIL PROTECTED] wrote: I am doing some testing using FOP (Flask Operator Panel) and so far, its going great! Been able to do callerid and also open a SugarCRM screen. All without having to install anything on the computer, just open a FOP browser screen and that's

[Asterisk-Users] asterisk integration with Quintum Tenor AXT800!

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all

[Asterisk-Users] Quintum Tenor AXT800!

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all

Re: [Asterisk-Users] IAX-IAX Trunking not works

6/8 6/3 00013ms 0001ms 0049ms gsm on another server shows test2*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/[EMAIL PROTECTED]/2192.168.0.77 adnan 2/20687 00026

[Asterisk-Users] IAX-IAX Trunking not works

Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low

[Asterisk-Users] iax trunking not works!

some solution for this Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] iax trunking not works!

connected to each other through IAX and if trunking not works its became a nightmare anyone suggest some other codecs except GSM ,i'll be very thankful. Thank You. Adnan Ahmed. On 5/13/05, Jay Milk [EMAIL PROTECTED] wrote: What codec are you using? -Original Message- From: Adnan

[Asterisk-Users] problems with Suse Linux Enterprise !

Hello, i am running suse linux enterprise edition of kernel version 2.6.5-7.97-smp, i have latest stable asterisk zaptel asterisk stuff compile fines i have TDM400P card with 1FXS and 3FXO modules, every time i probe with modprobe and issue ztcfg -vv commandit shows the following errors: also

[Asterisk-Users] want just few words from the list about SLES!

Hello, Sorry for bothering again i asked it early but noreply may be swap now ask it again hopefuly this time not vein my question is anyone try installing/running on Suse Linux Enterprise Server v9 ,may be vey helpful for me i try installing suse 9.2 professional but not successful not try on

[Asterisk-Users] problems with SLES!

Hello, sorry again i send a mail early which i can't receive so i send it again i am trying to install asterisk om suse linuex enterprise server but can;t make it also try udev settings but not working can anyone successfully installed asterisk on SLES helps a lot. Thanks in advance.

[Asterisk-Users] Asterisk's on Suse Linux Enterprise Server(SLESv9)

hello, can anyone installing/configuring asterisk's on SLES9 if someone can share his/her views experiences . Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] astguiclient error!

Hello, can anyone using astgui client i have a problem in installation phase everytime i try to create database from MySQL_AST_CREATE_tables.sql it gives error in phone table ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version

[Asterisk-Users] Connecting Multiple Asterisk Servers!

for guiding the best scalable robust setup because in future may be this setup expands so we also take a look at it any suggestions,tips,guidelines,weblinks may be very helpful. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] Connecting Multiple Asterisk Servers!

William, Thanks friend for your reply we have following setup: ipphones--routerasterisk server with channel bank loaded with Quad T1 card, also on channel bank several analog phones connected we deploy these setup at our headoffice and other branch offices situated in different cities we use

[Asterisk-Users] fxo card not workin in susev9.2!

Hello, I am new in linux and also suse i have a fxo card but its not working the errors are: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Notice: Configuration file is /etc/zaptel.conf line 143: Unable to open

Re: [Asterisk-Users] i am missing something!

thanks for replying but no change at all any other tips,suggestions thanks in advance On Wed, 9 Mar 2005 01:44:41 -0600, Jay Milk [EMAIL PROTECTED] wrote: You'll need canreinvite=no to each sip section in sip.conf, if you want * to stay in the loop. -Original Message- From: Adnan

[Asterisk-Users] i am missing something!

Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from

[Asterisk-Users] video confrencing

including video support . Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] Wrong CVS version ?

you are compiling in wrong sequence first zaptel then asterisk and after that asterisk-addons . hope this helps ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

Re: [Asterisk-Users] VoIP with Asterix

Dana Olson wrote: -- voip-info.org On Mon, 31 Jan 2005 20:21:51 -, Richard Dutton [EMAIL PROTECTED] wrote: Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk

just finish it if anyone like mysql go for it or someone love postgresql its ok but don't ruin the purpose of this list keep out these kind of mess sorry areski for that and thanks for your great work ___ Asterisk-Users mailing list

Re: [Asterisk-Users] ASTCC

Bilal Ghayad wrote: Dear Sebastian; Thanks a lot for your kindly advise to use ASTCC. But can u advise me the link for ASTCC to download it and wether it is open source (to download the source and work on it? Regards Bilal _ check it out http://www.voip-info.org/wiki-ASTCC regards

[Asterisk-Users] problems with astcc

;ip addr of sip phone disallow=all allow=gsm nat=yes qualify =1000 [adnan.007] username=adnan.007 type=peer secret=secret host=iptel.org disallow=all allow=gsm nat=yes qualify=1000 extensions.conf [general] static=yes writeprotect=no #include = /var/lib/astcc/astcc-exten.conf [incoming] exten = _N.,1

[Asterisk-Users] is wiki drunk

is there any problem with wiki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RealTime Drivers Connectivity Error

) but can't works I am using latest CVS-Head kindly pointout my mistakes. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] Segmentation Fault (core dumped)

i am facing unusual and wiered error in asterisk using Realtime MYSQL driver . Asterisk runs well and smoothly with absoulutely no error or warning but everytime i power-on my sip-phone ,booting, initializes and then asterisk suddenly quit with the error. _*Segmentation Fault (core

[Asterisk-Users] help regarding ASTCC

but no help/documentation. Thanks Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

[Asterisk-Users] MYSQL_FRIENDS

Hello *'s, Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot find this option.On wiki i found this. To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS. This enables database definition of both IAX2 and SIP

[Asterisk-Users] asterisk at large

this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] calling card application

can anyone using/integrating modified-prepaid-application avaiable on wiki . if anyone kindly guided me. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] modified prepaid application

how can we integrate the modified-prepaid application with asterisk because when i compile the app_prepaid it gives bunch of errors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk With PostgreSQL

is highly appreciated. This is my second post on this issue no responce on first one so plz take a while . Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Workimg On PostgrSQL

actually i have facing difficulties to connect PostgreSQL from asterisk what files i'll change to properly setup PostgreSQL with Asterisk any help is highly appreciated. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] Asterisk's Empty Folder

like to install the sample + + configuration files (overwriting any + + existing config files), run: + + make samples+ kindly pointout what's wrong i am doing bocz i spend almost a day or above but all in vein. Thanks in Advance Adnan Ahmed

Re: [Asterisk-Users] Asterisk's Empty Folder

Jim Radford wrote: You need to do a: make install and then make samples to install sample conf files. Jim On Wed, 8 Dec 2004, Adnan Ahmed wrote: Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am

[Asterisk-Users] Contact me Asap!

Hello Khurram, This is adnan from EBS kindly contact me as soon as possible i'll contact you on your number but its almost busy every time. Other *'s users kindly forgive me because i have no option right now. ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] Can't Register!

in my sip.conf file. sip.conf [general] port=5060 bindaddr=192.168.10.189 context=sip disallow=all allow=gsm nat=1 [101] username=101 type=friend host=dynamic secret=xyz context=from-sip callerid=101 dtnfmode=rfc2833 canreinvite=no qualify=1000 [iptel] username=adnan.007 type=friend secret=123 host

[Asterisk-Users] SIP Problem!

User/ANRCall IDSeq (Tx/Rx) LagJitterBuffer 0 active SIP channel(s) Kindly pointout my mistakes/errors and helping me out. Any Help Is Highly Appreciated. Thanks in Advance. Adnan Ahmed

[Asterisk-Users] Re:SIP Problem

I am very thankful to you people for helping me as much i imagine but i still need your help, problem is that i am not be able to dial from my analog phone conected to fxs card to my sip phone i change my configs but still no result. sip.conf [general] port=5060 bindaddr=192.168.10.193 allow=all

Re: [Asterisk-Users] Newbie Question

Leo Salas wrote: I am just learing some Linux and have been able to setup Asterisk samples and channels fxo card on ch.1 and fxs on ch 4. I have an Internet Polycom phone to use to test to/from internet and 1 analouge phone connected to port 4 of Digium TDM-400 with appropriate cards installed

Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!

for me, wow how silly I felt! Mike I am using Debian it's not working for me any other thaughts,tips suggestions because now i am very exhausted with this error i am looking almost everyplace wiki google but no luck kindly helping me out. On Sun, 21 Nov 2004 23:23:49 +0500, Adnan Ahmed

Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!

Mike Dent wrote: Did you try the iptables -L as I suggested though? It's probably still present in Debian. Mike On Mon, 22 Nov 2004 02:47:53 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote: Mike Dent wrote: Dont get caught by the same thing which had me ripping my hair out! I had installed

[Asterisk-Users] I Am Missing Something Somewhere Somehow!

0 active SIP channel(s) kindly pointout my mistakes/errors and helping me out. I am searching wiki,google but no luck i am tried several configs but all in vein please please helping me out :-( . Thanks In Advance . Adnan Ahmed. ___ Asterisk-Users

Re: [Asterisk-Users] I Am Missing Something Somewhere Somehow!

el Flynn wrote: Adnan Ahmed wrote: hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. snip *clisip show peers Name/UsernameHost Mask Port Status 101/101

Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'

Jose Hernandez wrote: I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [EMAIL PROTECTED] asterisk]#

[Asterisk-Users] Configuring Asterisk As A Sip Server

Hello Group, I want to configure my Asterisk Server As a SIP is there any possibality.How i do that.Any help is highly appreciated. Thanks in advance. Regards Adnan . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] Unable to create channel of type Zap!

mistakes/errors and helping me out. Thanks In Advance . Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] unable to create channel of type Zap

= s,1,Dial,Zap/1 ;exten = s,1,Dial,Zap/4 in above two lines which one os appropriate i am trying both options but no result. [outgoing] exten = 021NXX,1,Dial/Zap/1/${EXTEN:1} kindly pointout my mistakes/errors and helping me out. Thanks In Advance . Adnan Ahmed

[Asterisk-Users] Asterisk Hanging!

hi, everytime i run asterisk and looking asterisk log i found following errors: parse error: No category context for line 96 of extensions.conf Requested contexts didn't get merged Also asterisk not run just initialize and freezes and the log shows above description. my configs are: zaptel.conf

[Asterisk-Users] can i call my local phone to IP phone or vice versa

Hello, I am using TDM400 with FXO and FXS modules is there any possibality to call my local PSTN phone to my ip phone or vice versa for what configuration i adopt and if not possible what's the right approach. Thanks in Advance. ___ Asterisk-Users

[Asterisk-Users] Asterisk's Fails to start!

hi, I am using TDM400 with FXS and FXO modules,everytime i run asterisk and looking asterisk log i found following errors: parse error: No category context for line 96 of extensions.conf Requested contexts didn't get merged Also asterisk not run just initialize and freezes and the log shows

[Asterisk-Users] FXO Module Error

by incorrect module parameters,including invalid IO or IRQ parameters. What is this meaning ? Hardware or Software problem?I don't know. Kindly helping me out. Thanks in advance. Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Trunk doesn't work Adit 600/T100P

. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit conf voip-pbx print config - -Cactus.lite configuration file -Created on 01/01/1999 at 00:02:49 for adnan -This file is valid

[Asterisk-Users] Customized Call Parking

Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah. ___ Asterisk-Users

[Asterisk-Users] ALSA help required !

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. --- [chan_alsa.so] = (ALSA Console Channel Driver) Apr 20 18:28:34

[Asterisk-Users] Alsa driver doesn't initialize

---BeginMessage--- ---BeginMessage--- I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. --- [chan_alsa.so] = (ALSA