similar to: cvs stable

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

Hi all, I have 2 question. First: I use dovecot (version 2.2.9) with mdbox mail format. When I run dsync tool with "mirror" or "backup" parameters my source and destination directory synchronize correctly but if I delete some messages in user mailbox, deleted messages does not synced to destination. For example : atif at domain.com path is /mail/domain.com/atif/ and its

Hi, I downloaded Asterisk using this command a couple of weeks ago... # cd /usr/src # export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot # cvs login # cvs checkout zaptel libpri asterisk Can someone tell me what I need to type in order to get the latest stabe version when I rebuild my server ? Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I've seen a few posts about this but no fix. Anyone able to help? Here's what I did: I configured a brand new machine with Redhat 9.0. I made sure that I had: bison cvs gcc kernel-source libtermcap-devel ncurses-devel newt-devel openssl1096b openssl-devel readline41 readline-devel zlib zlib-devel When I went to get Asterisk I did the following: cvs checkout zaptel libpri and

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

I am sorry to ask such a simple questions. I have been using Asterisk successfully for well over a year now on three servers. I was using CVS HEAD, and the last time I updated was sometime back in July. I decided to switch to the recent stable 1.0.3. I built zaptel, libpri and asterisk, and installed them in that order. All installations reported success. (I stopped asterisk before installing

Dear All I am using Asterisk 1.4.17 in a calling card application. Following description explains the usage: A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a channel for this ip (Lets call it Channel A). Asterisk answers the call and play IVRs first asking the PIN and then destination number in an AGI making use of radius server for authentication/authorization. Once done

Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro

Hello everyone, I''m wondering what the current status of hardware performance counter usability in Xen is. I see some old posts describing the diffculties of virtualizing hardware counters within dom0 and the domUs, but not much else. Have they been implemented or are they in the process of being implemented? Or are there no future plans for implementation? Any help would be

Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf:

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or