Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
Sorry to insist, but I still not have any solution. Does anybody have an
idea ?
Thanks!
2011/4/20 John Alexis kasteris...@gmail.com
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load
iax accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And
I have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
other softphones (which parameters are loaded from the mysql database
iaxfriends).
BUT, i cannot call other softphones. I always got a message in the log
saying Everyone is busy/congested at this time (1:0/0/1).
So, i don't know where is the probleme : is it from iax accounts loaded
from the database, or the new account 444 ???
Below are the conf files and verbose output.
Thank you very much for your help :)
-
- iax.conf
-
[general]
bindport=4569
delayreject=yes
language=fr
autokill = yes
calltokenoptional = 0.0.0.0/0.0.0.0
minregexpire = 60
maxregexpire = 500
mohsuggest=default
careinvite=no
rtcachefriends=yes
[444]
type=friend
host=dynamic
context=special
secret=iop
-
- extconfig.conf:
-
[general]
[settings]
iaxusers = mysql,asterisk,iaxfriends
iaxpeers = mysql,asterisk,iaxfriends
voicemail = mysql,asterisk,voicemail
-
- Mysqldump from iaxfriends
-
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES
('admin.my.domain','friend','100','admin@my.domain','123','','default','','dynamic','10.0.100.56','26564','','','','','0','','','','','','en','admin.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES ('alice.my.domain','friend','111','admin@my.domain
','','alice@my.domain','456','','default','','dynamic','10.0.100.221','42478','','','','','1303301760','','','','','','en','alice.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
-
- extensions.conf:
-
[general]
[externe]
exten = 555,1,Dial(IAX2/111)
exten = 555,n,Hangup()
[special]
exten = 111,1,Dial(IAX2/111)
exten = 111,n,Hangup()
[default]
exten = 444,1,Dial(IAX2/444)
exten = 444,n,Hangup()
- Sip.conf (SIP server):
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
-
- Logs server:
-
-- Accepting AUTHENTICATED call from 10.0.100.238:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (),
priority = mine
-- Executing [111@special:1] Dial(IAX2/444-436, IAX2/111) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [111@special:2] Hangup(IAX2/444-436, ) in new stack
== Spawn extension (special, 111, 2) exited non-zero on 'IAX2/444-436'
-- Hungup 'IAX2/444-436'
-- Accepting AUTHENTICATED call from 10.0.100.50:
requested format = ulaw,
requested prefs = (),
actual format = gsm,
host prefs = (gsm|ulaw|alaw),
priority = mine
-- Executing [444@default:1] Dial(IAX2/alice.my.domain-8277,
IAX2/444) in new stack
-- Called 444
-- Call accepted by 10.0.100.238 (format gsm)
-- Format for call is gsm
-- IAX2/444-4734 is ringing
-- IAX2/444-4734 answered IAX2/alice.my.domain-8277
-- Hungup 'IAX2/444-4734'
== Spawn extension (default, 444, 1) exited non-zero on
'IAX2/alice.my.domain-8277'
-- Hungup 'IAX2/alice.my.domain-8277'
--
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[asterisk-users] Warm Transfer in Asterisk
Is there a way do what is sometimes called a 3rd party transfer in Asterisk. That is; Call A comes in and is answered B. B then places A on hold and calls C. After C answers, BC chat for a moment, then B brings A on line. After making intro's B then drops off call. Thanks, Jeff NeturallySpeaking -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warm Transfer in Asterisk
On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson jjohn...@neturallyspeaking.com wrote: Is there a way do what is sometimes called a 3rd party transfer in Asterisk. That is; Call A comes in and is answered B. B then places A on hold and calls C. After C answers, BC chat for a moment, then B brings A on line. After making intro’s B then drops off call. Yes it is called an attended transfer. You can use the atxfr feature code or most phones will have transfer capability built in. On Polycom phones the transfer button defaults to attended transfer. There is a separate blind transfer button as well. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot call to my server with SIP
Op 22-04-11 22:58, Steve Edwards schreef: Op 22-04-11 18:13, Eric Wieling schreef: sip set debug on should help On Fri, 22 Apr 2011, Paul van der Vlis wrote: I've tried it, but no, nothing... Sounds like you have very basic network issues. Can this host ping your SIP endpoint? Can this host ping any other host on your network? Can this host ping any host out on the Internet like 8.8.8.8? I am testing with accounts on public services (ekiga.net and sip2sip.info). I cannot run test from this machines. But I think my network is OK, you can try yourself to ping to my machine xen8.vandervlis.nl. I would like to have some test-tool 'sudo tcpdump' and sudo tcpdump port sip' may shed some clues. It's am idea, but letting my firewall log is something like this. With regards, Paul van der vlis. -- http://www.vandervlis.nl/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warm Transfer in Asterisk
I'll give it a whirl, Thanks, Jeff -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Saturday, April 23, 2011 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Warm Transfer in Asterisk On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson jjohn...@neturallyspeaking.com wrote: Is there a way do what is sometimes called a 3rd party transfer in Asterisk. That is; Call A comes in and is answered B. B then places A on hold and calls C. After C answers, BC chat for a moment, then B brings A on line. After making intro's B then drops off call. Yes it is called an attended transfer. You can use the atxfr feature code or most phones will have transfer capability built in. On Polycom phones the transfer button defaults to attended transfer. There is a separate blind transfer button as well. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot call to my server with SIP
Op 22-04-11 23:49, Jamie A. Stapleton schreef:
I can see your server just fine...
-bash-3.2# ./svmap.py xen8.vandervlis.nl
| SIP Device | User Agent | Fingerprint |
--
| 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|
However, if I try to call, Asterisk is saying:
-- Called p...@vandervlis.nl
[2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite:
Failed to authenticate on INVITE to ...;tag=as131f7b6a'
Ah, this is very good information. I see you, but I don't understand why
I don't see myself when I try this. Maybe my sip client (Ekiga) is not OK.
Asterisk log:
[Apr 22 23:46:50] NOTICE[29497] chan_sip.c: Sending fake auth rejection
for device Jamie A. Stapleton
sip:2233440...@sip2sip.info;tag=0wqaLzsAyMQwTdfcP2r0mG2FkPBQjEQF
Firewall log:
Apr 22 23:46:50 xen8 kernel: [3824476.043190] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=1320 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=1300
Apr 22 23:46:50 xen8 kernel: [3824476.043556] FW:IN= OUT=eth0
SRC=91.198.178.28 DST=81.23.228.150 LEN=782 TOS=0x00 PREC=0x00 TTL=64
ID=17809 PROTO=UDP SPT=5060 DPT=5060 LEN=762
Apr 22 23:46:50 xen8 kernel: [3824476.048153] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=411 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=391
What do you have allowguest
(http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?
I was testing security. It's like this:
sip.conf:
---
[general]
context=default
allowguest=no
alwaysauthreject=yes
(...)
[guests]
context=default
allowguest=yes
[trunk]
context=dialout
(...)
[phone-paul]
context=dialout
(...)
[phone-ann]
context=dialout
(...)
---
extensions.conf:
-
[default]
include = users
[dialout]
include = users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)
[users]
exten=6001,1,Dial(SIP/paul,20)
exten=6002,1,Dial(SIP/ann,20)
(...)
Thanks for your help!
With regards,
Paul van der Vlis.
--
http://www.vandervlis.nl/
--
_
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[asterisk-users] DTMF not being sent ( RFC2833 )
Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk 1.6 current with centos ( os that asterisk is developed on from my understanding ) with all default settings ( excluding logger.conf, dialplan and outboundproxy ) and I am having problems with the DTMF. Both servers were installed with CentOS 5.5 and were updated last night, after which I reinstalled asterisk. This did not resolve the issue. I am at wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files
Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
Hi, Using DumpChan(); Seems that Channel (where the call goes first) is a sub-channel of Context/Extension (where the call goes on CONNECT) ?? first I have: Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2: Then after: Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;1: Help ? On 23 April 2011 17:20, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance I'm a little confused by It Seems I cannot re-use this var in extension _888XX in context agentsOf course you can use it...but if you set bla to a different value in your code where your callfile is processed, Asterisk will (rightfully so) just set bla = to whatever you set it to Now, if the callfile doesn't send a channel through the context that you're trying to set blah, that's a little odd... Now, as far as retrieving the information about the interface that answered the calllook in queues.conf.samplethere's a nifty configuration option: *setinterfacevar=no ; (the default is no)* That option, when set to yes, causes several variables to be created *just*prior to the caller being bridged with the queue member... -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARA table definitions (1.8.*)
Where would one find, or better yet determine from code, all of the table definitions for ARA dynamic families? There seems to be some bits and pieces in various places around the internet, ie. voip-info, the definitive guide, ect. but nothing complete or definitive. I have wondered about this for years. Ideally we would have a script packaged with asterisk source, that could be run and would parse the source and generate the table create scripts, including all table columns, and save to file. We could then go in and customize the script from there, adding or removing columns as needed, ect. Thanks, Jason -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA table definitions (1.8.*)
On Sat, 2011-04-23 at 10:52 -0700, Jason Rogers wrote: Where would one find, or better yet determine from code, all of the table definitions for ARA dynamic families? There seems to be some bits and pieces in various places around the internet, ie. voip-info, the definitive guide, ect. but nothing complete or definitive. I have wondered about this for years. Ideally we would have a script packaged with asterisk source, that could be run and would parse the source and generate the table create scripts, including all table columns, and save to file. We could then go in and customize the script from there, adding or removing columns as needed, ect. Thanks, Jason -- I have to agree that info is scattered many places not just the asterisk site, but also voip-info, and a couple of threads on this list. One this i just noticed, is that all of them are related to 1.6 or older. That is, the field definitions for ip-adresses are just 15 characters wide. For most of us, that will be enough, but if you ever tried to store a ipv6 address into it, it will be severely truncated. So in order to be v4/v6 agnostic, it should be atleast 40 characters wide (4*8 hex charecters with seven colons)... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Huawei K3765 + Internet + SMS + Telephone
Hello asterisk asterisk, Am 2011-04-23 06:24:36, hacktest Du folgendes herunter: Look at this wiki for help. http://wiki.e1550.mobi/doku.php For asterisk, you can use your USB stick for voice/SMS but not internet at the same time. A separate internet connect is required per my understanding. Normaly the USB stick provide the UMTS/HSPA channel, but it is blocked by Asterisk and I do not know why. Maybe because asterisk try to reserve it vor Video-Telephony? Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux itsystems@tdnet France EURL itsystems@tdnet UG (limited liability) Owner Michelle KonzackOwner Michelle Konzack Apt. 917 (homeoffice) 50, rue de Soultz Kinzigstraße 17 67100 Strasbourg/France 77694 Kehl/Germany Tel: +33-6-61925193 mobil Tel: +49-177-9351947 mobil Tel: +33-9-52705884 fix http://www.itsystems.tamay-dogan.net/ http://www.flexray4linux.org/ http://www.debian.tamay-dogan.net/ http://www.can4linux.org/ Jabber linux4miche...@jabber.ccc.de ICQ#328449886 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot call to my server with SIP
If you don't see the call coming in when you have sip debug enabled, then the call is not making it to your server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis Sent: Friday, April 22, 2011 4:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cannot call to my server with SIP Op 22-04-11 18:13, Eric Wieling schreef: sip set debug on should help I've tried it, but no, nothing... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nat=yes
check this http://www.voip-info.org/wiki/view/Asterisk+sip+nat On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don’t require it) can cause problems? Regards and thanks in advance, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 codec selection and video
check this url, let me know if any problem http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+videobest On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies davies...@gmail.com wrote: Hi, Can anyone let me know how I can enable video (h.263) on SIP, but if a video call is passed over IAX, it will remove the video and pass the audio only. What I tried was: SIP - videosupport=yes - disallow=all - allow=alaw - allow=h263 IAX - disallow=all - allow=alaw What appears to occur is that the SIP call negotiates h263 video, and when passed over IAX, the h263 frames are passed, and are also accepted at the far end which also does not have a video codec allowed. Should that be happening? This is with 1.6.2.18-rc1. Am I missing a setting somewhere? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
