similar to: dialplan AGI DTMF

Does anyone configure that or is that possible ? Thanks in advance -- Best regards Vlad

Today morning cvs server checkout problem: cvs server: Updating asterisk-addons/format_mp3 cvs server: failed to create lock directory for `/usr/cvsroot/asterisk-addons/format_mp3' (/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied cvs server: failed to obtain dir lock in repository `/usr/cvsroot/asterisk-addons/format_mp3' cvs [server aborted]: read lock failed -

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

On oh323.conf I have: codec=G711U frames=20 But while connecting it gives me in log: ? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result: ? Table: ? ? G.723.1(5.3k){hw} <1> ? Set: ? ? 0: ? ? ? 0: ? ? ? ? G.723.1(5.3k){hw} <1> Which I don't have, so the connection is dropped. Any known solutions? (remote side has g711 u-Law) -- Witold Kr?cicki (adasi) adasi

HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast

This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng > Sent: Tuesday, August 10, 2004 8:35

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.

There is option automon => *1 in features.conf As I understand when *1 pressed during conversation => recording should begin. But unfortunately it doesn't work for me. I use CVS-HEAD-01/27/05 Does anyone has that feature working? Thanks -- Best Regards VladK

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Hi! I tried to use Parking Calls I use Asterisk 13, but I can't park any calls and it returns me [Feb 3 16:56:11] WARNING[1693]: pbx.c:12543 ast_context_verify_includes: Context 'ramais' tries to include nonexistent context 'parkedcalls' What is the correct code for put in extensions.conf? Can be this example below? [parkedcalls] exten => 700,1,ParkedCall(701) exten

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf ------ context=inbound-analog callgroup=2 channel=2 ------ sip.conf ------ [ciscok] type=friend

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes

Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the

I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thank?s Excuse my English Joao Carlos Moura

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,