similar to: Can't get video on one server of 4

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

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

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

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

Hi, I tried to turn on Video and get the following cli-WARNING output -- Executing [8600 at outgoing-kamailio:1] Answer("PJSIP/7000-00000000", "") in new stack > 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 -- Executing [8600 at outgoing-kamailio:2] ConfBridge("PJSIP/7000-00000000", "8600") in new stack --

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that "friend", as follows:   [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer   Unfortunately the accountcode for the calls originating from "mypeer" doesn't show up in the log (either CSV or ODBC). All the other

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

Hi all Maybe somebody has an idea. I'm tracing a very strange phenomena... I've a connection from Asterisk to a SIP PBX. Most calls have a caller ID. Some International calls don't have any. Now it looks like those calls without caller ID never get to the context where incomming calls from this SIP PBX should get to.... Examples: Call with Caller ID: (slightly anonymized)

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

Hi In Asterisk 1.6/realtime Mysql, we can't put a username/password in a Dial Command ?: 'Dial', 'SIP/Username:Password at MYPEER/${EXTEN},180,r' Thanks Olivier

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be desired. Case in point -- if you compare the

Anybody have any experience with setting up dhcpd in failover mode between two servers? I set this up on a couple of servers, and it seems to be working, but I don't think it is working "right". It appears both servers are replying to all requests (which for renewals works okay because they both give the same address, but new requests get two different responses). I thought that

Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both having an extension [115], one as type peer (caller side 1.4) and one as friend (callee side 1.8). Phones from both location connect to Asterisk from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret

Dear sir, I got trouble on InnoMedia video phone with Asterisk-1.2.4. If InnoMedia video phone as a caller, then the call will be a success, no any problem. The problem happens: If InnoMedia video phone to be a callee, the call can not make successfully. For instance, Caller 23267668 dialed to callee 23267663, callee 23267663 was ringing. If I picked up the callee's phone, then abnormal

Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensi?n 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100