similar to: ast_writefile: No such format 'h261', yet h261 is the only video format that works.

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've encountered a problem playing back a .wav file to an Ekiga client: My dialplan looks like: exten => 730,1,answer exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign) exten => 730,n,hangup Sovereign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays:

Dear Folks, [Test_Context] exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _911.,2,Set(CALLERID(num)=xxxxxxx) exten => _911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)}) exten => _911.,5,Set(${CALLERID}=${CALLERID(num)}) exten =>

Hello all, I am trying to leave a video voicemail but am unable to do so. I am using Ekiga (formerly Gnomemeeting) to make a SIP connection to Asterisk 1.2.4. Ekiga supports h261 for video. The call connects and negotiation seems okay. When I leave a message, however, only the audio is recorded. Looking in the log file afterwards I see many messages like this: Mar 21 22:02:34 WARNING[2418]

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

Hello, I have been attempting to get the Monitor function to accept a loal variable substitution in order to use the same filename later in the same context. Monitor does not appear to like it, as it attempts to use wav|filename as the recording type, as opposed to just wav. Here is what I get if I just supply a filename directly (it works fine): --context----------------------------- exten

I'm trying to get two Windows Messenger clients to communicate with video and audio though asterisk. I'm running into one of two problems. I get garbled audio under the current config. I had another config where I could get a voice call to work but using video would cause the caller to get music on hold. (very odd) Calling a phone hanging off of an TDM the audio works great. This is

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi All... I'm trying to use the record() app and it complains that it can't open it's file because permission was denied. I'm running the released Asterisk on Debian Linux. The target directory is workd writable. Here is the relevant part of the dialplan: exten => 1,1,Playback(leave-message) exten => 1,2, Record(/var/local/whois-messages/whois-${contactid}:wav|6|120)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

I just set up a WRT54GS and now I can't dial out or in. sip show registry shows: CODE: SELECT ALL Host dnsmgr Username Refresh State Reg.Time draytel.org:5060 N xxxxx 120 Request Sent I seemed to recall that running in cli always showed a response back, but there's nothing now. Using

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323 But I'm getting all kinds of errors about PWLIB... I built using the newest PWLIB and OpenH323 from CVS Error log from make below make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES

I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323 But I'm getting all kinds of errors about PWLIB... I built using the newest PWLIB and OpenH323 from CVS Error log from make below make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES

Following is the errors when I tried to compile oh323 in FreeBSD 5.3. Asterisk is updated from cvs. asterisk# gmake for x in wrapper asterisk-driver; do gmake -C $x build || exit 1 ; done make: illegal option -- - usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags] [-E variable] [-f makefile] [-I directory] [-j max_jobs] [-m directory] [-V variable]

When i attempt to make asterisk-oh323 I get the following error. Iam using the following versions : asterisk-oh323-0.6.3b openh323-v1_13_5 pwlib-v1_6_6 openh323_1.13.5-make.patch redhat 8 kernel 2.4.18-14 Anyone can help to get a way out ? Or suggetions for compatible versions ? -----------------------------------------------------------------------------------

Hello.... Asterisk is compiled and running perfectly... But when i try to compile Channel_h323... that's another story :-|. I'm compiling using RH9, OpenH323_1.12.2, pwlib_1.5.2, i compiled and installed them myself... GnuGk is running smoothly using them. Also my intention is to terminate calls from a SIP client through Asterisk and into my GnuGK... has anyone accomplished this ?