similar to: FAX

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI; ############################ #To read asterisk variable values. $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse();

Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/default at extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00XXXXXX at PSTN GAteway Asterisk server is connected succeffully to database...As soon as i make a call i got the following error message:

Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

I'm trying to test out Speex for our branch to branch connections, but am running in to a problem. I downloaded the Speex source code for 1.2rc1, did a ./configure, make and make install then went to my asterisk folder did a ./configure, make clean make menuconfig verified that speex is enabled, saved config then did make, stopped asterisk then make install and start asterisk. Did the

Hi all! I would like to make a service with recording sounds and playing back to caller. I had wrote the script but it failed at Read statement with file not found error. I have put some file test into script and this is what happen on verbose level 9. -- Executing [8298 at default:8] Record("DAHDI/27-1",

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can