similar to: 2 same sip extension number on 2 asterisk - call not passing on certain condition

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

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

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, 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 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

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 have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel

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 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

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

On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote: > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer <peername>" for a phone. > bpi*CLI> sip show peer 0049177xxxxxxx > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| >

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

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

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 : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon