similar to: VOIP-INFO down?

Hi to all, I have a question about the callerid (msn number) off calls comming FROM a queue. This is my setup: - ISDN using zap (zaphfc) - Incomming calls arrive in a queue - One of the members of the queue is my cell phone... member => Zap/g1/XXX73XX19 (X to protect my privacy ;-)) The problem is the call from the queue to my cell phone. In a dailplan i can set the outgoing msn using

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

So it looks like Livevoip went Bankrupt ------------------------------------------- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

When an agent logs into a queue using AgentCallBackLogin, he should be ready to take calls until he logs out right? For some reason the first time a customer calls the queue, it rings the agent just fine but after the agent hangs up the phone and the next caller calls the queue, no more calls will be transferred to the agent. He shows as logged in, but the calls wait in the queue forever and

[root@asterisk zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy [root@asterisk

Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP

Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 572/0x23C) (Terminator) > Message type: ALERTING (1) >

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

I don't know if there is some fault in compiling or a bug of the new R-1.7.0 version: cl.pam.2 <- pam(as.dist(1-cor(mel.data)),2) plot(cl.pam.2) perform a right partitioning and silhouette plot on the old R-1.6.2 instead "Error in clusplot.default(x$diss,...... ; x is not numeric" is the output on the new R-1.7.0. Same platform: RH8.0 i686. Some suggestions? A.S.

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0

Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but nothing is working. I even installed BIND on the asterisk box ...that didn't even work. Once I pull the plug on the internet, I cant dial anything. John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax:

Hello all, i am new to shorewall and i already have a question ;) i am running a mailserver in my dmz (or actually this will be when = evertything will be working fine with shorewall) with public ip = addresses.. i have a subnet of 8 ip addresses (255.255.255.248 mask) and = i was planning of the classic 3 nic (eth0-2) setup... the dmz should = work with proxy-arping...=20 now my quesion is