similar to: Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?

Thanks to the great support at Rhino Equipment and Digium, this has finally been solved. I wanted to post the solution back to the list in case anyone else is having a similiar issue. I started by calling Rhino support so I could eliminate channel bank configuration as the issue. We were able to determine the channel bank and signalling were all working as expected. I then began to monitor

Hi, I have host running 2.6.16 dom0 kernel (Gentoo based). All my domUs are using 2.6.16 custom compiled kernels. Now my question is - is it possible to use newer version for domUs, for example 2.6.20 FC6 domU kernel? BR Peter Braun _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Hello, I?m using virt-install to build a guest system with CentOS. The system boots up, times out with dracut-initqueue timeout and drops me into an emergency shell. If I exit the shell the install continues and I get a working machine. Any ideas? virt-install \ -n TEST \ -r 8192 \ --os-type=linux \ --disk=/vm-images/test.img,device=disk,bus=virtio,size=100,format=raw \

Hi All This mail is from a new comer to asterisk , just exploring I was going through the digium web site , In need some advice and suggestions , which card I should chose , should I choose TE411P (with on board echo cancellation ) or TE410P If I go for TE411P , I will be spending 1,000 $ more and for me finance part is critical , will I face

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

Hello, I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I installed the following -rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0- beta2.tar.gz -rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0- beta1.tar.gz -rw-r--r-- 1 root root 80019 Sep 21 13:25 libpri-1.4.0-beta1.tar.gz -rw-r--r-- 1 root root 1523413 Sep 21 13:25

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

I''ve been trying to match multiple public IPs in an ingress qdisc. The idea is to allow these specific IPs and aggregate value of 256 kbits incoming to the interface. Can anyone tell me how this can be effectively done if at all possible with tc? Thanks in advance -- Corey Rogers <jrog@sunbeach.net> _______________________________________________ LARTC mailing list /

I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything (recompile zaptel+asterisk+spandsp ; echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing worked. The only solution that worked for me was to install and use NVFaxDetect. HTH.

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Hi I can provide my customers one or several phone lines by using an ATA through the SIP protocol. Is there a similar box that would allow me to provide a PRI (23B +D) to a customer using SIP ? Thanks Patrick

Hi, In Asterisk, what happens to the files when both legs of the call hangs up? Is there a way to create a conference room on the flight? i.e. without pre-defining the conference ID in meetme.conf. Thnx much. -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2872 bytes Desc: not available Url :

Hi all, 1. Just bought a new TE411P and about to install it replacing the existing TE410P. I am assuming I need to set echocancel=no and echocancelwhenbridged=no now that it will be done in hardware, correct? 2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart from being on a call)? 3. The Digium webite says "By supporting 16ms with 128 channels or

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

I just wanted to see if any one else has seen this or could help point me in the right direction on this problem. I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I

At Sangoma we do quite a lot of back-to back T1 and E1 connections. T1 is not a very fussy connection, as the baud rate is only about 750 kbps. In our experience, for error free communications you can use the following rules of thumb: Up to 50 ft: Flat patch cable Up to 500 ft: Ordinary twisted telephone cable Cat 5 may be overkill unless you are going hundreds of feet. David Yat Sin