similar to: 1.0.6 music on hold bug ?!

Hello, Are there any T1/E1 PCMCIA cards available on the market supported by zaptel drivers and asterisk ? I need to make some demos at my clients with asterisk and it's a pain to move around with a midi-tower computer just for that. Thanks, Calin.

Hello people, It seems that my * does not react to tos=<whatever> field in iax.conf. I am using latest CVS HEAD code. Can anybody help me with this issue? ps: if i go to chan_iax2.c and modify the initial definition of tos variable, it works fine marking packets with the value specified there: static int tos=16; if i put random text in iax.conf's tos=, chan_iax2 refuses to load

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin.

Hello everybody! I am having the following problem and since I am a beginner in playing with asterisk, i can't solve it: I am trying to integrate my existing H.323 network in real world telephony by ISDN cards. The problem is that i DON'T want to change all e164 numbers in my h.323 network and my ISDN provider doesn't accept those identities (CIDs). So, i have to spoof the outgoing

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom "Hold" button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays.

Hi everyone! I am using OpenSuse 10.3 with desktop effect compiz-fusion and wine 0.9.57! I have installed counter strike 1.6 with wine and everything went fine. When i run the game with no desktop effect works fine but when i run with desktop effects it doesn`t start and it tells me: Code: calin at DarkStar:~/.wine/drive_c/Program Files/Valve> wine hl.exe -game cstrike

Hi, my name is Calin and I''m new to linux, but I guess its the right place to ask this: what do I set on a linux RH9 box with 2.4.24 kernel to route a 10 machine private network (192.168.x.x) by 3 limited bandwidth, public IPs (193.231.x.x). The network uses a switch, the linux box has 1 ethernet card, the link is available trough a wireles ethernet bridge from my ISP. I begun to read

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

Hi, all I am trying to install E100P card, the 'modprobe zaptel' is ok, but when I did 'modprobe wct1xxp', I got such error, so can not load the driver for the card. /lib/modules/2.4.20-8/misc/wct1xxp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts

Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp 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 Cisco configuration: isdn switch-type primary-qsig