similar to: FW: Problem with Asterisk PRI forwarding to SER

Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I have been using * for a couple of month and now I wanted to install the Asterisk Calling Card. But when I try to install I get the following error: [root@terraserver1 astcc]# make install mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi

Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Hi all, Is there a way of disabling ALERTING message on a PRI channel? I have a problem .* is sending ALERTING message to the Nortel telco switch of my local provider BEFORE it dial the number it has to .if the number is busy or invalid there is no way we can tell this to the switch because it has already been told that the phone is ringing I am using asterisk Asterisk CVS-04/06/04-10:46:21 with

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Hi All, the * is working rigth for incoming calls and internal calls, but when trying to call out we got hanged up. The hangup reason is AST_CAUSE_INVALID_IE_CONTENTS I've been searching in the mailing list archive as I thing that some thing similar happens to someone else but did not find. We are runnig asterisk 1.2.4 extensions.conf [default] exten =>

I have SER running with Asterisk, both on seperate servers. If I call another SIP number from my SIP phone SER looks up the phone number to see if it's online. If it's not online it forwards the call to Asterisk. How do I configure the extensions.conf file so that calls being forwarded to Asterisk destined for VoiceMail do not conflict with normal outbound calls destined for the PSTN?

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

Hi folks. My client is wanting to use call forwarding configured on their phones (Linksys SPA942), with a PRI from their provider. When we configure call forwarding, we invariably get a "The number you have dialed is not in service" message from the providers. Examining the detailed dial plan debugging as well as the PRI debugging, the number is dialed correctly. The only difference

> On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: >> Only way I have managed to get Zap channel to reject a call on TE110P >> without answering it, is to dial number that is not handled in dialplan >> (I >> have a ISDN PRI with 100 number DID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >>

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

I am trying to use rsync to backup from a site we will call "office" and another we will call "home." Both sites have DSL accounts provided by Arachnet. At present the files being backed up don't all all to be backed up, but OTOH we wish to backup lots more files that aren't being backed up now. First, we create a local backup on our office machine which happens to

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear I am using ser + asterisk, for setting up land line calling. only probelm, each unregistered soft phone can places the call only with callerid, this is critical problem, because any number(soft phone) , has a limit time to use this system, best Mani ____________________________________________________________________________________Be a better Globetrotter. Get better travel

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1">Hi,<br> <br> i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull

Dear R People: Do any of the linear model or regression function calculate the Durbin-Watson test statistic for serial correlation, please? I found the hat matrix, studentized residuals, and so on, but no D-W. Thanks so much! Sincerely, Erin M. Hodgess, Ph.D. Associate Professor Department of Computer and Mathematical Sciences University of Houston - Downtown One Main Street Houston, TX 77002