similar to: moh/playback for non-zap interfaces

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, This new version has more options (account code, AMA flags, lib tracing) in the config file and some improvements in the build process. The code is available from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael Manousos

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those cards provide some reliable timing. So I have no

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

This module wont compile can anyone give me any assistance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040614/03ae433c/attachment.htm

Hello all, This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio" problem of the previous release. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

Believe it or not, wine runs the Install Shield program perfectly when installing Lotus Notes. However, when creating mail, after I type one letter (doesn't matter what it is) every letter after it dissappears until you hit backspace, delete, or click the mouse in the window. This is annoying as some of us *like* to see what we're typing in an email :-) I'm Running RedHat 6.2 and the

Hi all, I'm not sure if this is the correct mailing list for this (I was going to send to Asterisk-Biz, but seems more for this one). Anyway, I'm having more problems with Broadvoice. I still can't get calls unless I comment out the secret= line in sip.conf, but now I can't even place test calls to it from my cell phone. When I do, I get a fax macine. Debugging SIP shows NO

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303