similar to: can we install 10 PCI card on asterisk

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. any help appericiated Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I''ve written a minimal sort of Perl module that dynamically creates and destroys traffic control rules for specific IPs. I''m currently using it for a user bandwidth control application at a client site. The module essentially gets Ethernet device(s), IP address and in/out speeds as input and dynamically creates classes, queues

hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice

Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session?

Hi, I have a client connected to the ''net through 3 ISP''s. Have set up a Linux box to do routing and load sharing for the 3 connections. A fourth interface is connected to the LAN with private IP addresses. Am using iptables to SNAT traffic to the appropriate IP depending on the interface the packet gets routed onto. The setup looks something like this: Interface IP

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

#!/bin/bash # checksetexternip.sh # Author: John Cahill email at johncahill.net # Licence: GPL v3 # Description: script that queries checkip.dyndns.com to find the server's external IP address. Updates asterisk's externip value and does a sip reload if necessary. # Last modified 06/02/2012 is_ip(){ input=$1 octet1=$(echo $input | cut -d "." -f1) octet2=$(echo $input

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

dear all, what is the meaning of this *Got SIP response 603 "Declined" back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this regards -------------- next part -------------- An HTML attachment was scrubbed... URL: