similar to: What kind of configuration do I need to run Asterisk ?

Hi all, I realise that asterisk's codec negotiation has been discussed in the past multiple times. What I haven't been able to understand is how asterisk decides which video codecs to advertise to the other end when canreinvite=no in sip.conf and the initial caller doesn't support video. My tests are quite simple, I use an asterisk with 4 peers all on the same LAN. My sip.conf

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour

Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : anthony at mbdsys.com Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

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

Hello on Behalf Of idont know, Sangoma has a Media Gateway solution via SS7. They I believe are the only ones capable of connecting Asterisk via SS7. You may want to check them out. Heidi -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of idont know Sent: April 6, 2006 10:29 AM To: asterisk-biz@lists.digium.com

Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing

Hi all, I am using to Xlite to save video voice mail. when i retreive it, then only video show , no voice is here out. Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box. I did following configuration In Sip.conf videosupport=yes [phone1] type=friend host=dynamic context= employees mailbox=101 at default

Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? 2. From package description, is this package enough or not to allow transcoding with G711 ? For instance, in the following

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've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

I want to make a recording server. Were people can record a message, then we take the wav file, and move it to another server. I'm going to have a Toll Free DID directing to my * server. Right off the bat I will usualy be getting 12 concurrent calls at a time. And am planning on getting up to 100 in the near future. So using the G711 Codec, recieving 100 calls through a DID. What kind of

I can create an audio call file and specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course), and I do specify also the Codecs: h264,h263 etc... Asterisk reports: *File /tmp/video does not

Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks,

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/8005555555 at outbound.vitelity.net Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable:

By browsing on the mailing list I learned that its possible to generate .h263 asterisk friendly files with gstreamer. The script below it's supposed to do just that, however I get error when trying it out locally. gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux ! ffdec_h263 ! videoscale ! video/x-raw-yuv,width=352,height=288 ! ffenc_h263 rtp-payload-size=512 !

Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your