[asterisk-users] Trunking betweeb two Asterisk System

[Faraj Khasib]

Hi guys,
I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 
1.6
but I cannt make it work, can any body help me plz?
Thank you
--
_
-- 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] Executing Script after MixMonitor is called

[Faraj Khasib]

Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is 
called but it doesnt execute the command, I tried the command in terminal it 
worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame -b 8 -t -F 
-m m --bitwidth 8 --quiet /var/spool/asterisk/monitor/${CALLFILENAME}.wav 
/var/spool/asterisk/monitor/${CALLFILENAME}.mp3  rm -f 
/var/spool/asterisk/monitor/${CALLFILENAME}.wav)
exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)

--
_
-- 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] Change the caller's phone number

[Faraj Khasib]

try the following 
Set(${CALLERID}=722979797 722979797)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com]
Sent: Thursday, January 19, 2012 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Change the caller's phone number

Hi,
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew the call came from
our system.
But I have a problem with that, not all phone number change some of the
anonymous calls stay anonymous and the recipient See in the caller ID
display unlisted number.
I use this commend:
Set(CALLERID(all)=722979797 722979797)

Is anyone having a similar problem or know what the problem?

Thanks.


--
_
-- 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] Change the caller's phone number

[Faraj Khasib]

or this 

Set(${CALLERID(all)}=722979797 722979797)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Thursday, January 19, 2012 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Change the caller's phone number

try the following 
Set(${CALLERID}=722979797 722979797)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com]
Sent: Thursday, January 19, 2012 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Change the caller's phone number

Hi,
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew the call came from
our system.
But I have a problem with that, not all phone number change some of the
anonymous calls stay anonymous and the recipient See in the caller ID
display unlisted number.
I use this commend:
Set(CALLERID(all)=722979797 722979797)

Is anyone having a similar problem or know what the problem?

Thanks.


--
_
-- 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

--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

I already tried what u posted  didnt work 
but thanx for the reply :)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind 
[govoi...@gmail.com]
Sent: Wednesday, January 04, 2012 11:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call type in dial plan

Hi,
Sorry for late reply. Hope you've already found out something about it.

What version of asterisk you are using, that function for choosing 
inbound/outbound call leg codecs is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
https://issues.asterisk.org/view.php?id=13243

Regards,
Sammy


On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
thats excatly what I want, can u plz give me the command, I want to choose only 
ulow

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Sammy Govind [govoi...@gmail.commailto:govoi...@gmail.com]
Sent: Tuesday, January 03, 2012 3:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call type in dial plan

Hi,

For such call you just need to select the outbound codec before the dial() app.

choose the audio-only codecs and thus no video codec strings will be exchanged 
in that call.

--
Regards,
Sammy

On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.commailto:fkha...@iconnecths.commailto:fkha...@iconnecths.com
 wrote:
this is what my SIP Invite message when I make Video call

INVITE 
sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102mailto:sip%3A6500@192.168.21.102mailto:sip%253A6500@192.168.21.102
 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
From: 
sip:6097@192.168.21.102mailto:sip%3A6097@192.168.21.102mailto:sip%3A6097@192.168.21.102mailto:sip%253A6097@192.168.21.102;tag=1857098215
To: 
sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102mailto:sip%3A6500@192.168.21.102mailto:sip%253A6500@192.168.21.102
Contact: 
sip:6097@192.168.21.193:52933;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
CSeq: 324677463 INVITE
Content-Type: application/sdp
Content-Length: 588
Max-Forwards: 70
Route: sip:192.168.21.102:5060;lr;transport=udp
Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: Medcor
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 192.168.21.193
s=-
c=IN IP4 192.168.21.193
t=0 0
m=audio 36372 RTP/AVP 8 0 9 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
m=video 59296 RTP/AVP 125 106 121 103
a=rtpmap:125 VP8/9
a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:106 H264/9
a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; 
max-mbps=11880
a=rtpmap:121 MP4V-ES/9
a=fmtp:121 profile-level-id=3
a=rtpmap:103 H263-1998/9
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2

when I make Audio call requests I dont have the video part  but at receiver 
since two clients can make video call they have Asterisks adds the Video Part 
in request sent to receiver,I dont want that part added , how I can delete it ?
--
_
-- 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


--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video its 
sent to the other client as video call .I tried settings the 
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- 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

--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

I tried also in asterisk 1.8 setting outbound variable  but didnt work also 

https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different 
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source for 
the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video its 
sent to the other client as video call .I tried settings the 
SIP_CODED_INBOUND and outbound also ... but no luck 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- 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

--
_
-- 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

--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- 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

--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

didnt work also :(

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new 
stack
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin
g codec to 'gsm' for this call because of ${SIP_CODEC} variable
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin
g ${SIP_CODEC} variable because it is not shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin
g codec to 'gsm' for this call because of ${SIP_CODEC} variable
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin
g ${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do
es not exist in any format
[Jan  4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco
me (format 0x4 (ulaw)): No such file or directory
[Jan  4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil
e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n
ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to video 
call request (since my other end has also all codecs)
my app clients can do Audio and Video call,
now the Video call is ok
but the Audio part get converted to video request ...so I am trying to limit 
the codec to only audio codec...

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end
Ignoring ${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-,
2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new
stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500)
in n ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

Any suggestion will be great 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to video 
call request (since my other end has also all codecs)
my app clients can do Audio and Video call,
now the Video call is ok
but the Audio part get converted to video request ...so I am trying to limit 
the codec to only audio codec...


--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

there is nothing in sip.conf about what u asked
but 6500 is a queue with following info
[6500]
fullname = testing
strategy = rrmemory
timeout = 15
wrapuptime = 15
autofill = no
autopause = no
joinempty = yes
leavewhenempty = no
reportholdtime = no
maxlen = 0
musicclass = test
member = SIP/6251
member = SIP/6252
member = SIP/6253
member = SIP/6254

now the user 6251 is a user with following info and caller 6000

[6000]
username = 6000
transfer = yes
mailbox = 6000
call-limit = 100
type = peer
fullname = 6000
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
callcounter = yes

[6251]
username = 6251
transfer = yes
mailbox = 6251
call-limit = 100
type = peer
fullname = 6251
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6251
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Please post the sip.conf entries for 6000 and 6500.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to
video call request (since my other end has also all codecs) my app clients
can do Audio and Video call, now the Video call is ok but the Audio part get
converted to video request ...so I am trying to limit the codec to only
audio codec...

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end Ignoring
${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

allow=all

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

What about the allow/disallow lines in sip.conf?

--
_
-- 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] Set Call Codec in extension.conf

[Faraj Khasib]

yup  and video support is yes

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Both sides?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

allow=all

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

What about the allow/disallow lines in sip.conf?

--
_
-- 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

--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

this is what my SIP Invite message when I make Video call

INVITE sip:6500@192.168.21.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
From: sip:6097@192.168.21.102;tag=1857098215
To: sip:6500@192.168.21.102
Contact: 
sip:6097@192.168.21.193:52933;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
CSeq: 324677463 INVITE
Content-Type: application/sdp
Content-Length: 588
Max-Forwards: 70
Route: sip:192.168.21.102:5060;lr;transport=udp
Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: Medcor
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 192.168.21.193
s=-
c=IN IP4 192.168.21.193
t=0 0
m=audio 36372 RTP/AVP 8 0 9 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
m=video 59296 RTP/AVP 125 106 121 103
a=rtpmap:125 VP8/9
a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:106 H264/9
a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; 
max-mbps=11880
a=rtpmap:121 MP4V-ES/9
a=fmtp:121 profile-level-id=3
a=rtpmap:103 H263-1998/9
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2

when I make Audio call requests I dont have the video part  but at receiver 
since two clients can make video call they have Asterisks adds the Video Part 
in request sent to receiver,I dont want that part added , how I can delete it ? 
--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

thats excatly what I want, can u plz give me the command, I want to choose only 
ulow

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind 
[govoi...@gmail.com]
Sent: Tuesday, January 03, 2012 3:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call type in dial plan

Hi,

For such call you just need to select the outbound codec before the dial() app.

choose the audio-only codecs and thus no video codec strings will be exchanged 
in that call.

--
Regards,
Sammy

On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
this is what my SIP Invite message when I make Video call

INVITE sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
From: sip:6097@192.168.21.102mailto:sip%3A6097@192.168.21.102;tag=1857098215
To: sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102
Contact: 
sip:6097@192.168.21.193:52933;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
CSeq: 324677463 INVITE
Content-Type: application/sdp
Content-Length: 588
Max-Forwards: 70
Route: sip:192.168.21.102:5060;lr;transport=udp
Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: Medcor
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 192.168.21.193
s=-
c=IN IP4 192.168.21.193
t=0 0
m=audio 36372 RTP/AVP 8 0 9 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
m=video 59296 RTP/AVP 125 106 121 103
a=rtpmap:125 VP8/9
a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:106 H264/9
a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; 
max-mbps=11880
a=rtpmap:121 MP4V-ES/9
a=fmtp:121 profile-level-id=3
a=rtpmap:103 H263-1998/9
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2

when I make Audio call requests I dont have the video part  but at receiver 
since two clients can make video call they have Asterisks adds the Video Part 
in request sent to receiver,I dont want that part added , how I can delete it ?
--
_
-- 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

[asterisk-users] How to make SIP guest call

[Faraj Khasib]

Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

Actaully I didnt find a good example how to configure the guest call in 
asterisk other than allowGuest in SIP.conf, anybody have a good example for 
that?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call

Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

thank you very much for this explanation, but my question does my client have 
to be registered first, right? what do i Use to register ... there should be 
information to register with using guest,
I got your idea about the security, and I can work with that ... but at cleints 
I need to have information to log with?right? 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn 
[da...@klaverstyn.com.au]
Sent: Tuesday, January 03, 2012 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Hi,

The point of SIP guest calls is that there is no username and password required 
to make calls.  If you have enabled guest calls then whatever extensions you 
have allowed in the allocate default sip context anyone will be able to dial.

If you have in your sip.conf file

context=from-vsp; Default context for incoming calls
allowguest=yes  ; Allow or reject guest calls (default is yes)

and in your extensions.conf file

exten = 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3)
exten = 202,n,Set(CALLERID(NAME)=Guest SIP User)
exten = 202,n,Dial(SIP/202,30,r)
exten = 202,n,VoiceMail(202@default,us)
exten = 202,n,HangUp


... then anyone will be able to call 202.

The key is to make sure people cannot make trunk calls from the guest context.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Tuesday, 3 January 2012 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Actaully I didnt find a good example how to configure the guest call in 
asterisk other than allowGuest in SIP.conf, anybody have a good example for 
that?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call

Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_
-- 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

--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

for example if I am using x-lite as client, how to I connect as guest from 
client ...I am allowing guests at asterisk server

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

thank you very much for this explanation, but my question does my client have 
to be registered first, right? what do i Use to register ... there should be 
information to register with using guest,
I got your idea about the security, and I can work with that ... but at cleints 
I need to have information to log with?right?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn 
[da...@klaverstyn.com.au]
Sent: Tuesday, January 03, 2012 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Hi,

The point of SIP guest calls is that there is no username and password required 
to make calls.  If you have enabled guest calls then whatever extensions you 
have allowed in the allocate default sip context anyone will be able to dial.

If you have in your sip.conf file

context=from-vsp; Default context for incoming calls
allowguest=yes  ; Allow or reject guest calls (default is yes)

and in your extensions.conf file

exten = 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3)
exten = 202,n,Set(CALLERID(NAME)=Guest SIP User)
exten = 202,n,Dial(SIP/202,30,r)
exten = 202,n,VoiceMail(202@default,us)
exten = 202,n,HangUp


... then anyone will be able to call 202.

The key is to make sure people cannot make trunk calls from the guest context.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Tuesday, 3 January 2012 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Actaully I didnt find a good example how to configure the guest call in 
asterisk other than allowGuest in SIP.conf, anybody have a good example for 
that?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call

Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_
-- 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

--
_
-- 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

--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

anyone?
what should x-lite account be for guest user ?I tried guest but didnt work

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

for example if I am using x-lite as client, how to I connect as guest from 
client ...I am allowing guests at asterisk server

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

thank you very much for this explanation, but my question does my client have 
to be registered first, right? what do i Use to register ... there should be 
information to register with using guest,
I got your idea about the security, and I can work with that ... but at cleints 
I need to have information to log with?right?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn 
[da...@klaverstyn.com.au]
Sent: Tuesday, January 03, 2012 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Hi,

The point of SIP guest calls is that there is no username and password required 
to make calls.  If you have enabled guest calls then whatever extensions you 
have allowed in the allocate default sip context anyone will be able to dial.

If you have in your sip.conf file

context=from-vsp; Default context for incoming calls
allowguest=yes  ; Allow or reject guest calls (default is yes)

and in your extensions.conf file

exten = 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3)
exten = 202,n,Set(CALLERID(NAME)=Guest SIP User)
exten = 202,n,Dial(SIP/202,30,r)
exten = 202,n,VoiceMail(202@default,us)
exten = 202,n,HangUp


... then anyone will be able to call 202.

The key is to make sure people cannot make trunk calls from the guest context.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Tuesday, 3 January 2012 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

Actaully I didnt find a good example how to configure the guest call in 
asterisk other than allowGuest in SIP.conf, anybody have a good example for 
that?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call

Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_
-- 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

--
_
-- 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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Registering multi-clients

[Faraj Khasib]

hey all,
My problem is that I am trying to have multiclients call my SIP queue, now each 
client is not authorized  so I tried to make them call using the same 
extension but I got call overlap between all clients, now what I want is a way 
that I can make all my client call the SIP queue using SIP protocol, I am 
thinking using guest Call it would solve my problem, can u plz help me, if you 
have any other suggestion plz do
Thanx
--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

thank you for your reply, but x-lite cannt dail without an active account  
dail is disabled without any account
My problem is that I am trying to have multiclients call my SIP queue, now each 
client is not authorized  so I tried to make them call using the same 
extension but I got call overlap between all clients, now what I want is a way 
that I can make all my client call the SIP queue using SIP protocol, I am 
thinking using guest Call it would solve my problem, can u plz help me, if you 
have any other suggestion plz do
Thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists 
[asterisk-l...@puzzled.xs4all.nl]
Sent: Tuesday, January 03, 2012 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

On 03-01-12 14:13, Faraj Khasib wrote:
 anyone?
 what should x-lite account be for guest user ?I tried guest but didnt work

A guest does not need an account on your asterisk server so you do not
need to configure an account on xlite. Instead on xlite you just dial
extension@ip_of_your_asterisk_server

Regards,
Patrick

--
_
-- 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] How to make SIP guest call

[Faraj Khasib]

thanx alot ... :) that helped 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Roland 
[aster...@rolandow.com]
Sent: Tuesday, January 03, 2012 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

I managed to do that once by using another SIP account, for example at 
Voipbuster. It's free. Once you are connected, you can still use ext@IP of your 
server. I guess you could use any other free SIP account.

On Tue, Jan 3, 2012 at 4:01 PM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
thank you for your reply, but x-lite cannt dail without an active account  
dail is disabled without any account
My problem is that I am trying to have multiclients call my SIP queue, now each 
client is not authorized  so I tried to make them call using the same 
extension but I got call overlap between all clients, now what I want is a way 
that I can make all my client call the SIP queue using SIP protocol, I am 
thinking using guest Call it would solve my problem, can u plz help me, if you 
have any other suggestion plz do
Thanx

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Patrick Lists 
[asterisk-l...@puzzled.xs4all.nlmailto:asterisk-l...@puzzled.xs4all.nl]
Sent: Tuesday, January 03, 2012 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call

On 03-01-12 14:13, Faraj Khasib wrote:
 anyone?
 what should x-lite account be for guest user ?I tried guest but didnt work

A guest does not need an account on your asterisk server so you do not
need to configure an account on xlite. Instead on xlite you just dial
extension@ip_of_your_asterisk_server

Regards,
Patrick

--
_
-- 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


--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib
--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

Please help, I have tried many things I cannt make it work, when I make an 
audio call it is converted by asterisk to video call request, Please how to set 
the call type at extensions.conf, I tried setting the codec manually but didnt 
work also... any help .. any suggest will be great
Thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Monday, January 02, 2012 3:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call type in dial plan

Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib
--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

I use asterisk 1.6, my clients are sip clients, I dail using audio call in my 
clients but the request is recieved at the other client as video call request 
since I am enabling video support for sip

Sent from my iPhone

On ٠٢‏/٠١‏/٢٠١٢, at ١١:٤٩ م, Doug Lytle supp...@drdos.info wrote:

 
 Faraj Khasib wrote:
 Please help, I have tried many things I cannt make it work, when I make an 
 audio call it is converted by asterisk to video call request
 
 Not that I can help, since I don't do any video calling.
 
 But, if you don't give any information about your system (OS and 
 version, Asterisk version and what type of phone you are using), you're 
 not likely to get much of a response.
 
 Doug
 
 
 -- 
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.
 
 
 --
 _
 -- 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] Set Call type in dial plan

[Faraj Khasib]

Which is?! What I am missing how to set dail plan in extension.conf to pass 
call type as its  Not convert request to video

Sent from my iPhone

On ٠٣‏/٠١‏/٢٠١٢, at ٧:٢٩ ص, virendra bhati 
virbh...@gmail.commailto:virbh...@gmail.com wrote:

Hi,

Please give you sip phone name and sip.conf and extensions.conf details which 
is using for that communication.
And CLI output of asterisk is also required.


On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my 
clients but the request is recieved at the other client as video call request 
since I am enabling video support for sip

Sent from my iPhone

On ٠٢‏/٠١‏/٢٠١٢, at ١١:٤٩ م, Doug Lytle 
supp...@drdos.infomailto:supp...@drdos.info wrote:


 Faraj Khasib wrote:
 Please help, I have tried many things I cannt make it work, when I make an 
 audio call it is converted by asterisk to video call request

 Not that I can help, since I don't do any video calling.

 But, if you don't give any information about your system (OS and
 version, Asterisk version and what type of phone you are using), you're
 not likely to get much of a response.

 Doug


 --
 Ben Franklin quote:

 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.


 --
 _
 -- 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



--

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer

--
_
-- 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] Set Call type in dial plan

[Faraj Khasib]

Here is the thing, my sip client can call the same. Extension once as audio and 
once as video, so I cannt turn off video supportat reciever, what I guess can 
be done is in extension.conf , there must be flag or something I can manipulate 
...
Sent from my iPhone

On ٠٣‏/٠١‏/٢٠١٢, at ٨:١٩ ص, virendra bhati 
virbh...@gmail.commailto:virbh...@gmail.com wrote:

Which is means like if you are using sip 1234 then give the details of [1234] 
into that open thread and relevent extensions details too

On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
Which is?! What I am missing how to set dail plan in extension.conf to pass 
call type as its  Not convert request to video

Sent from my iPhone

On ٠٣‏/٠١‏/٢٠١٢, at ٧:٢٩ ص, virendra bhati 
virbh...@gmail.commailto:virbh...@gmail.com wrote:

Hi,

Please give you sip phone name and sip.conf and extensions.conf details which 
is using for that communication.
And CLI output of asterisk is also required.


On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib 
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my 
clients but the request is recieved at the other client as video call request 
since I am enabling video support for sip

Sent from my iPhone

On ٠٢‏/٠١‏/٢٠١٢, at ١١:٤٩ م, Doug Lytle 
supp...@drdos.infomailto:supp...@drdos.info wrote:


 Faraj Khasib wrote:
 Please help, I have tried many things I cannt make it work, when I make an 
 audio call it is converted by asterisk to video call request

 Not that I can help, since I don't do any video calling.

 But, if you don't give any information about your system (OS and
 version, Asterisk version and what type of phone you are using), you're
 not likely to get much of a response.

 Doug


 --
 Ben Franklin quote:

 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.


 --
 _
 -- 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



--

Thanks and regards

 Virendra Bhati
+91-8885268942tel:%2B91-8885268942
Software Engineer

--
_
-- 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



--

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer

--
_
-- 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

[asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

Hi All,
I am trying to record Call, but when the call is done I have one file but the 
conversation inside it is separate into calls conversation and receiver  
its single file but separate recording,
How can I make it mixed together so the conversation will be normal?
Thanx
--
_
-- 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] Monitor Command Records separate channales

[Faraj Khasib]

I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command
this is how I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales

Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales

Hi All,
I am trying to record Call, but when the call is done I have one file but
the conversation inside it is separate into calls conversation and receiver
 its single file but separate recording, How can I make it mixed
together so the conversation will be normal?
Thanx
--
_
-- 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

--
_
-- 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] Monitor Command Records separate channales

[Faraj Khasib]

Asterisk 1.6.2 but sox I don't know but now it is the latest version, my 
problem is not mixing  It's the same file but inside that file two seperate 
records first callers then reciever

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:

 According to the monitor documentation, the format you specified should be
 calling SOX and mixing on call completion.  What versions of SOX and
 Asterisk are you using?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how I use
 Monitor
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Monitor Command Records separate channales
 
 Hi All,
 I am trying to record Call, but when the call is done I have one file but
 the conversation inside it is separate into calls conversation and receiver
  its single file but separate recording, How can I make it mixed
 together so the conversation will be normal?
 Thanx
 --
 _
 -- 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
 
 --
 _
 -- 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
--
_
-- 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] Monitor Command Records separate channales

[Faraj Khasib]

Can u plz tell me how , I forgot how to run asterisk cli

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:

 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 Asterisk 1.6.2 but sox I don't know but now it is the latest version, my 
 problem is not mixing  It's the same file but inside that file two 
 seperate records first callers then reciever
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
 
 According to the monitor documentation, the format you specified 
 should be calling SOX and mixing on call completion.  What versions of 
 SOX and Asterisk are you using?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how I 
 use Monitor 
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Monitor Command Records separate channales
 
 Hi All,
 I am trying to record Call, but when the call is done I have one file 
 but the conversation inside it is separate into calls conversation and 
 receiver  its single file but separate recording, How can I make 
 it mixed together so the conversation will be normal?
 Thanx
 --
 _
 -- 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
 
 --
 _
 -- 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
 --
 _
 -- 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

Re: [asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

My call happens with a queue , there is no full file but there is queue and 
queue is useless, can u give me unix command to search all log files and print 
moniter line?

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:

 Asterisk -vvvrc 
 Is how you would get it live
 After the fact you might find it in /var/log/asterisk/full
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 3:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 Can u plz tell me how , I forgot how to run asterisk cli
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
 
 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 Asterisk 1.6.2 but sox I don't know but now it is the latest version, 
 my problem is not mixing  It's the same file but inside that file 
 two seperate records first callers then reciever
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
 
 According to the monitor documentation, the format you specified 
 should be calling SOX and mixing on call completion.  What versions 
 of SOX and Asterisk are you using?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how 
 I use Monitor 
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%
 m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Monitor Command Records separate channales
 
 Hi All,
 I am trying to record Call, but when the call is done I have one file 
 but the conversation inside it is separate into calls conversation 
 and receiver  its single file but separate recording, How can I 
 make it mixed together so the conversation will be normal?
 Thanx
 --
 _
 -- 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
 
 --
 _
 -- 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

Re: [asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

I already searched using grep for the monitor word ... It doesn't exists

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١١:١٥ م, Faraj Khasib fkha...@iconnecths.com wrote:

 My call happens with a queue , there is no full file but there is queue and 
 queue is useless, can u give me unix command to search all log files and 
 print moniter line?
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
 
 Asterisk -vvvrc 
 Is how you would get it live
 After the fact you might find it in /var/log/asterisk/full
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 3:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 Can u plz tell me how , I forgot how to run asterisk cli
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
 
 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 Asterisk 1.6.2 but sox I don't know but now it is the latest version, 
 my problem is not mixing  It's the same file but inside that file 
 two seperate records first callers then reciever
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
 
 According to the monitor documentation, the format you specified 
 should be calling SOX and mixing on call completion.  What versions 
 of SOX and Asterisk are you using?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how 
 I use Monitor 
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%
 m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate 
 channales
 
 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj 
 Khasib
 Sent: Wednesday, December 28, 2011 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Monitor Command Records separate channales
 
 Hi All,
 I am trying to record Call, but when the call is done I have one file 
 but the conversation inside it is separate into calls conversation 
 and receiver  its single file but separate recording, How can I 
 make it mixed together so the conversation will be normal?
 Thanx
 --
 _
 -- 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
 
 --
 _
 -- 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] Monitor Command Records separate channales

[Faraj Khasib]

see attached ...

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales

Even using Queue there should still be a /var/log/asterisk/full that records 
the Monitor then the following Queue/Dial commands.  What is in your 
/var/log/asterisk?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales

My call happens with a queue , there is no full file but there is queue and 
queue is useless, can u give me unix command to search all log files and print 
moniter line?

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:

 Asterisk -vvvrc
 Is how you would get it live
 After the fact you might find it in /var/log/asterisk/full


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 3:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Can u plz tell me how , I forgot how to run asterisk cli

 Sent from my iPhone

 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:

 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Asterisk 1.6.2 but sox I don't know but now it is the latest version,
 my problem is not mixing  It's the same file but inside that file
 two seperate records first callers then reciever

 Sent from my iPhone

 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:

 According to the monitor documentation, the format you specified
 should be calling SOX and mixing on call completion.  What versions
 of SOX and Asterisk are you using?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how
 I use Monitor
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
 %
 m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Monitor Command Records separate channales

 Hi All,
 I am trying to record Call, but when the call is done I have one
 file but the conversation inside it is separate into calls
 conversation and receiver  its single file but separate
 recording, How can I make it mixed together so the conversation will be 
 normal?
 Thanx
 --
 
 _
 -- 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

Re: [asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales

I would wager that your setup dumps what would normally be in /v/l/a/full into 
/v/l/a/messages

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales

see attached ...

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales

Even using Queue there should still be a /var/log/asterisk/full that records 
the Monitor then the following Queue/Dial commands.  What is in your 
/var/log/asterisk?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales

My call happens with a queue , there is no full file but there is queue and 
queue is useless, can u give me unix command to search all log files and print 
moniter line?

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:

 Asterisk -vvvrc
 Is how you would get it live
 After the fact you might find it in /var/log/asterisk/full


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 3:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Can u plz tell me how , I forgot how to run asterisk cli

 Sent from my iPhone

 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:

 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Asterisk 1.6.2 but sox I don't know but now it is the latest version,
 my problem is not mixing  It's the same file but inside that file
 two seperate records first callers then reciever

 Sent from my iPhone

 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:

 According to the monitor documentation, the format you specified
 should be calling SOX and mixing on call completion.  What versions
 of SOX and Asterisk are you using?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how
 I use Monitor
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
 %
 m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same?
 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales

 Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

It got stuck ...

Sent from my iPhone

On ٢٨‏/١٢‏/٢٠١١, at ١١:٢٩ م, Danny Nicholas da...@debsinc.com wrote:

 Try
 # grep 'onitor' /var/log/asterisk/messages
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 3:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 but i tiried these commands and I didnt find anything about Monitor
 [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
 [root@c-24-1-71-68 asterisk]# grep -R 'monitor' * 
 
 From: asterisk-users-boun...@lists.digium.com 
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 3:23 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 I would wager that your setup dumps what would normally be in /v/l/a/full 
 into /v/l/a/messages
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 3:20 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 see attached ...
 
 From: asterisk-users-boun...@lists.digium.com 
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
 [da...@debsinc.com]
 Sent: Wednesday, December 28, 2011 3:18 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 Even using Queue there should still be a /var/log/asterisk/full that records 
 the Monitor then the following Queue/Dial commands.  What is in your 
 /var/log/asterisk?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
 Sent: Wednesday, December 28, 2011 3:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate channales
 
 My call happens with a queue , there is no full file but there is queue and 
 queue is useless, can u give me unix command to search all log files and 
 print moniter line?
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
 
 Asterisk -vvvrc
 Is how you would get it live
 After the fact you might find it in /var/log/asterisk/full
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 3:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales
 
 Can u plz tell me how , I forgot how to run asterisk cli
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
 
 Can you post a CLI output of the Monitor output?  I'm supposing that 
 something in your $(STRFTIME) string might be eating the M option.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales
 
 Asterisk 1.6.2 but sox I don't know but now it is the latest version,
 my problem is not mixing  It's the same file but inside that file
 two seperate records first callers then reciever
 
 Sent from my iPhone
 
 On ٢٨‏/١٢‏/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
 
 According to the monitor documentation, the format you specified
 should be calling SOX and mixing on call completion.  What versions
 of SOX and Asterisk are you using?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
 Khasib
 Sent: Wednesday, December 28, 2011 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Monitor Command Records separate
 channales
 
 I installed SOX( it was not installed before). Will that solve my problem?
 if not what are the parameter for the mixMonitor Command this is how
 I use Monitor
 exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
 %
 m
 -%d_%H
 :%M:%S)}_${SIP_HEADER(email)},m)
 is Mix Monitor will have the same

Re: [asterisk-users] Monitor Command Records separate channales

[Faraj Khasib]

I attached log, but there is nothing unusual in it ...all normal ...

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com]
Sent: Wednesday, December 28, 2011 4:06 PM
To: Faraj Khasib
Subject: Your message to asterisk-users awaits moderator approval

Your mail to 'asterisk-users' with the subject

RE: [asterisk-users] Monitor Command Records separate channales

Is being held until the list moderator can review it for approval.

The reason it is being held:

Message body is too big: 1004233 bytes with a limit of 40 KB

Either the message will get posted to the list, or you will receive
notification of the moderator's decision.  If you would like to cancel
this posting, please visit the following URL:


http://lists.digium.com/mailman/confirm/asterisk-users/7c086c1398347b43db2e2984127934cd8cbde5c4


--
_
-- 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] Asterisk Sip Media Call Type

[Faraj Khasib]

Hi all,
I am trying to make a SIP Video and Audio Call, Now when I add at the Asterisk 
the video Support and the right codec whether I make Audio or Video Call from 
my clients the Call will be received as Video Call, so the problem is if I make 
from one client Audio or Video Call it will be recieved as Video Call, Can you 
plz help me try to solve this problem? Where should I change the Call Media 
Type at Asterisks
Regards
Faraj Khasib
--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

thank you for ur solution, I did this in dail plan yesterday ... it took me 5 
hours to find that solution , I wish u replied to me earlier but thanx :)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Torbjörn Abrahamsson 
[torbjorn.abrahams...@gmail.com]
Sent: Monday, November 28, 2011 12:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Does Asterisk alter   the Headers of  
INVITE  Message

Well, It doesn't forward the INVITE at all, as asterisk is NOT a proxy. It
creates a totally new INVITE when you issue the Dial application, with its
own set of headers.

Now, you can pass the Test header with something like this (taken from
memory...):

SipAddHeader(Test: ${SIP_HEADER(Test)})

Do that prior to the call to the Dial application, and you will see your
header in the outgoing INVITE. Of course this means that your dial plan need
to know which headers to pass.

// T


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: den 28 november 2011 00:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE
Message

Any body knows how I can configure Asterisk SIP to pass all Header
Parameters?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Sunday, November 27, 2011 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk alter the   Headers of
INVITE  Message

thats my main question if u can see Does Asterisk alter the Headersof
INVITE  Message
I am using ASterisk NOW proxy  I didnt configure it to delete anything ,
Can u tell me how I can change it to pass that parameters?
thanx

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers   of
INVITE  Message

On 11/27/2011 05:25 PM, Faraj Khasib wrote:

 Yes, see attached ... Proxy server alter my Test custom header and
 delete it, Is there a way to include it in message sent from SIP
 Proxy to target?

That would be a proxy configuration issue, wouldn't it?

In principle, the proxy should be passing these messages through
unmodified, unless you have an explicit configuration directive that
instructs it to remove headers from the INVITE.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

--
_
-- 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

--
_
-- 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

--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e 
(email=m...@me.com) but when I check the Message at the target that header is 
not there
So I is Askterisk altering the Message and Is there away to include extra 
headers for SIP INVITE Message?
Thank u
--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

Please guys anybody knows How can I send a unique token to the Receiver at the 
Invite call? Is that possible?
 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Sunday, November 27, 2011 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Does Asterisk alter the Headers of INVITE Message

Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e 
(email=m...@me.com) but when I check the Message at the target that header is 
not there
So I is Askterisk altering the Message and Is there away to include extra 
headers for SIP INVITE Message?
Thank u
--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

I tried that with my SIP Cleint but the custom Header is not reaching the 
cleint ... Does the asketrisk delete that?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov 
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message

On 11/27/2011 04:27 PM, Faraj Khasib wrote:

 Please guys anybody knows How can I send a unique token to the
 Receiver at the Invite call? Is that possible?

Custom SIP headers are a common way to do that.  Try SIPAddHeader().

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

Yes, see attached ...
Proxy server alter my Test custom header and delete it, Is there a way to 
include it in message sent from SIP Proxy to target?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov 
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:19 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers ofINVITE  
Message

On 11/27/2011 04:53 PM, Faraj Khasib wrote:

 I tried that with my SIP Cleint but the custom Header is not reaching
 the cleint ... Does the asketrisk delete that?

Are you sure?  Have you taken a packet capture to confirm?

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

--
_
-- 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
Received (1128 bytes): 192.168.1.101:5060 - 192.168.1.104:50495
INVITE sip:6500@192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport
From: sip:6097@192.168.1.101;tag=194243250
To: sip:6500@192.168.1.101
Contact: 
sip:6097@192.168.1.104:50495;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511
CSeq: 947428168 INVITE
Content-Type: application/sdp
Content-Length: 257
Max-Forwards: 70
Route: sip:192.168.1.101:5060;lr;transport=udp
Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Test: testing
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: Medcor
Supported: 100rel

v=0
o=doubango 1983 678901 IN IP4 192.168.1.104
s=-
c=IN IP4 192.168.1.104
t=0 0
m=audio 38378 RTP/AVP 8 0 9 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15



Transaction(id='z9hG4bK1421826827' method=INVITE server=true) created.

Sending (263 bytes): 192.168.1.101:5060 - 192.168.1.104:50495
begin
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495
To: sip:6500@192.168.1.101
From: sip:6097@192.168.1.101;tag=194243250
Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511
CSeq: 947428168 INVITE
Content-Length: 0

end


Transaction(id='z9hG4bK1421826827' method=INVITE server=true) Transaction 
timeout Timer started, will triger after 9.

Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE 
server=false) created.

Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE 
server=false) Timer A(requst retransmit timer) started, will triger after 500.

Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE 
server=false) Timer B(calling state timeout timer) started, will triger after 
32000.

Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE 
server=false) Transcation timeout timer started,timeout after 18 ms

Sending (1199 bytes): 192.168.1.101:5060 - 192.168.1.101:59495
begin
INVITE sip:6500@192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.101;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb;rport
Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495
From: sip:6097@192.168.1.101;tag=194243250
To: sip:6500@192.168.1.101
Contact: 
sip:6097@192.168.1.104:50495;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511
CSeq: 947428168 INVITE
Content-Type: application/sdp
Max-Forwards: 69
Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Test: testing
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,NOTIFY,PRACK,UPDATE,REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: Medcor
Supported: 100rel
Record-Route: sip:192.168.1.101;lr
Content-Length: 257

v=0
o=doubango 1983 678901 IN IP4 192.168.1.104
s=-
c=IN IP4 192.168.1.104
t=0 0
m=audio 38378 RTP/AVP 8 0 9 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
end


Received (421 bytes): 192.168.1.101:5060 - 192.168.1.101:59495
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 
192.168.1.101;rport;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb
From: sip:6097@192.168.1.101;tag=194243250

Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

thats my main question if u can see Does Asterisk alter the Headersof  
INVITE  Message
I am using ASterisk NOW proxy  I didnt configure it to delete anything , 
Can u tell me how I can change it to pass that parameters?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov 
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers   of  INVITE  
Message

On 11/27/2011 05:25 PM, Faraj Khasib wrote:

 Yes, see attached ... Proxy server alter my Test custom header and
 delete it, Is there a way to include it in message sent from SIP
 Proxy to target?

That would be a proxy configuration issue, wouldn't it?

In principle, the proxy should be passing these messages through
unmodified, unless you have an explicit configuration directive that
instructs it to remove headers from the INVITE.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

--
_
-- 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] Does Asterisk alter the Headers of INVITE Message

[Faraj Khasib]

Any body knows how I can configure Asterisk SIP to pass all Header Parameters?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Sunday, November 27, 2011 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk alter the   Headers of  INVITE  
Message

thats my main question if u can see Does Asterisk alter the Headersof  
INVITE  Message
I am using ASterisk NOW proxy  I didnt configure it to delete anything , 
Can u tell me how I can change it to pass that parameters?
thanx

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov 
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers   of  INVITE  
Message

On 11/27/2011 05:25 PM, Faraj Khasib wrote:

 Yes, see attached ... Proxy server alter my Test custom header and
 delete it, Is there a way to include it in message sent from SIP
 Proxy to target?

That would be a proxy configuration issue, wouldn't it?

In principle, the proxy should be passing these messages through
unmodified, unless you have an explicit configuration directive that
instructs it to remove headers from the INVITE.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

--
_
-- 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

--
_
-- 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] I want to participate in development

[Faraj Khasib]

Hi Asterisks Developers,
I want to learn all about IP telephony and I was wondering If I can participate 
in development?
Thank you

--
_
-- 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] Does Asterisk Support SIP Video Call ?

[Faraj Khasib]

Hi all,
I tried making a video SIP call using Asterisk  But it didnt workonly 
voice call works?
Regards
Faraj Khasib
--
_
-- 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] Does Asterisk Support SIP Video Call ?

[Faraj Khasib]

Now I did, thank you for ur help and it works :D

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI 
[ad...@tootai.net]
Sent: Wednesday, November 16, 2011 5:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk Support SIP Video Call ?

Le 16/11/2011 10:23, Faraj Khasib a écrit :
 Hi all,
 I tried making a video SIP call using Asterisk  But it didnt workonly 
 voice call works?

Hi Faraj,

Asterisk support H261, H263, H263+ and H264. Video calls are working
since at least 1.4 version. You have to activate it by setting
videosupport=yes in sip.conf

--
Daniel

--
_
-- 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

[asterisk-users] Multiple SIP endpoint registrations

[Faraj Khasib]

Hi guys,
I want to ask if its possible to make calls using one SIP account,
The problem is like this : I have an iPhone app and I want all my users to call 
the same extension which is virtual extension to my call center,
so the iPhone app will be using the same SIP account for all users
lets say for example: 
iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center)
Now My question is about the iPhone user part... Does the Asterisk 1.8 support 
that all my iPhone users register with the same account(6000@mydomain) and call 
that extension(dont worry about this extension)?
Regards
Faraj Khasib
--
_
-- 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] Multiple SIP endpoint registrations

[Faraj Khasib]

I have phone system and I am connecting Asterisk to it trunk.
Now I want my iphone users (clients ) to call my call center which is in phone 
system by using the same SIP account 
the user will call asterik with for example 6000 as account then the asterik 
will forward the call via trunk to that Phone system.
My question is this :
Can all my iPhone users which are using the 6000 as an account call the call 
center ? with asterisk 1.7?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming 
[kpflem...@digium.com]
Sent: Tuesday, November 15, 2011 8:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple SIP endpoint registrations

On 11/15/2011 07:28 AM, Faraj Khasib wrote:
 Hi guys,
 I want to ask if its possible to make calls using one SIP account,
 The problem is like this : I have an iPhone app and I want all my users to 
 call the same extension which is virtual extension to my call center,
 so the iPhone app will be using the same SIP account for all users
 lets say for example:
 iPhone users uses 6000@mydomain to call 9000@my domain(which is the call 
 center)
 Now My question is about the iPhone user part... Does the Asterisk 1.8 
 support that all my iPhone users register with the same 
 account(6000@mydomain) and call that extension(dont worry about this 
 extension)?

No Asterisk does not support multiple registrations to the same SIP
account (AoR), but that is irrelevant in this case, because
registrations are not used for placing calls *to* Asterisk, only
receiving calls *from* Asterisk.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

--
_
-- 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] Multiple SIP endpoint registrations

[Faraj Khasib]

btw the call is one direction from clients to Call center 
My question can be rephrased  can I make call without registration to an 
registered SIP account?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Tuesday, November 15, 2011 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple SIP endpoint registrations

I have phone system and I am connecting Asterisk to it trunk.
Now I want my iphone users (clients ) to call my call center which is in phone 
system by using the same SIP account
the user will call asterik with for example 6000 as account then the asterik 
will forward the call via trunk to that Phone system.
My question is this :
Can all my iPhone users which are using the 6000 as an account call the call 
center ? with asterisk 1.7?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming 
[kpflem...@digium.com]
Sent: Tuesday, November 15, 2011 8:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple SIP endpoint registrations

On 11/15/2011 07:28 AM, Faraj Khasib wrote:
 Hi guys,
 I want to ask if its possible to make calls using one SIP account,
 The problem is like this : I have an iPhone app and I want all my users to 
 call the same extension which is virtual extension to my call center,
 so the iPhone app will be using the same SIP account for all users
 lets say for example:
 iPhone users uses 6000@mydomain to call 9000@my domain(which is the call 
 center)
 Now My question is about the iPhone user part... Does the Asterisk 1.8 
 support that all my iPhone users register with the same 
 account(6000@mydomain) and call that extension(dont worry about this 
 extension)?

No Asterisk does not support multiple registrations to the same SIP
account (AoR), but that is irrelevant in this case, because
registrations are not used for placing calls *to* Asterisk, only
receiving calls *from* Asterisk.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

--
_
-- 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

--
_
-- 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