[asterisk-users] realm question

[Hans Witvliet]

Hi all,

Trying to find where i got wrong in my config

Is the "realm" parameter in sip.conf only used for possible
autentication?

The thing is, i got my box more-or-less working as i wanted,
but i can only reach internal functions (like echo-test and so on) and
other sip-clients if i dial "1234@fqdn", while i was expected to be able
to just dial "1234"

I presume i have either a mismatch between how the softphones register,
and my asterisk conf.

Kind regards, Hans


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Re: [asterisk-users] Blind Transfer Connected

[Olivier]

2011/7/5 Nikhil 

> Hi all
>In asterisk if blind transfer failed ,call is not connecting back .
>
> For Eg:
>A make call to B through asterisk,then B transfer the call to C. If C
> did not answer the call ,A  and B Call should connect back.
>
IMHO, blind tranfer definition is to NOT connect A and B back


> But this is not happening with asterisk(A and B call is disconnecting).
>
> Does anyone knows about this?
>
> Thanks
> Nikhil
>
>
>
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[asterisk-users] SIP Presence not working

[Deka, Rajib IN MAA SL]

Hello all,

I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is 
not working properly for all users.
Our SIP client sends SIP:SUBSCRIBE to all the configured extensions in asterisk 
during registration process. Asterisk replies with 200 OK for all SUBSCRIBE. 
But if I run "sip show subscriptions" in CLI prompt, it shows only a few live 
subscriptions per user. The result is not consistent; sometime it shows 
subscription status for all the extensions and sometime a few (per user). We 
have allowsubscribe=yes and callcounter=yes in sip.conf file.

Can somebody please help me to debug this issue and identify the root cause?

Regards,
Rajib


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[asterisk-users] Blind Transfer Connected

[Nikhil]

Hi all
In asterisk if blind transfer failed ,call is not connecting back .

For Eg:
A make call to B through asterisk,then B transfer the call to C. If 
C did not answer the call ,A  and B Call should connect back.But this is 
not happening with asterisk(A and B call is disconnecting).


Does anyone knows about this?

Thanks
Nikhil



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[asterisk-users] DTMF between sip trunks and PRIs

[James Lamanna]

Hi,
I'm looking for some advice on how to solve DTMF issues.
I have 2 boxes, one which is the connection to the PSTN (PSTN) through
PRIs and SIP trunks, and a second (PBX) which has UAs registered to
it.
We have a customer that has an existing pbx that we trunk analog lines
to using a GXW-4008.
The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF.

The issue I'm currently having is with inbound DTMF.
PBX and PSTN are connected through a standard sip trunk. Both machines
are on the same physical switch.

Here are the results I've seen:

PBX <-> PSTN using rfc2833 | Incoming call on PRI  | DTMF on pbx
voicemail system fails (dup/missing digits)
PBX <-> PSTN using inband | Incoming call on PRI  | DTMF on pbx
voicemail system is correct

PBX <-> PSTN using rfc2833 | Incoming call on SIP  | DTMF on pbx
voicemail system is correct
PBX <-> PSTN using inband | Incoming call on SIP  | DTMF on pbx
voicemail system is correct

All asterisk versions are 1.4.35.
PRI card is a Sangoma A104 with HW DTMF detection.

Does asterisk just have a problem converting the DTMF from the
D-channel to rfc2833?
The DTMF log looks ok (I dialed '642'), so I'm not sure where the
issue is coming in.


[Jul  4 21:05:44] DTMF[9769] channel.c: DTMF begin '6' received on Zap/15-1
[Jul  4 21:05:44] DTMF[9769] channel.c: DTMF begin passthrough '6' on Zap/15-1
[Jul  4 21:05:44] DTMF[9769] channel.c: DTMF end '6' received on
Zap/15-1, duration 100 ms
[Jul  4 21:05:44] DTMF[9769] channel.c: DTMF end accepted with begin
'6' on Zap/15-1
[Jul  4 21:05:44] DTMF[9769] channel.c: DTMF end passthrough '6' on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF begin '4' received on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '4' on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF end '4' received on
Zap/15-1, duration 100 ms
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF end accepted with begin
'4' on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF end passthrough '4' on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF begin '2' received on Zap/15-1
[Jul  4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '2' on Zap/15-1
[Jul  4 21:05:46] DTMF[9769] channel.c: DTMF end '2' received on
Zap/15-1, duration 100 ms
[Jul  4 21:05:46] DTMF[9769] channel.c: DTMF end accepted with begin
'2' on Zap/15-1
[Jul  4 21:05:46] DTMF[9769] channel.c: DTMF end passthrough '2' on Zap/15-1

Thanks.

-- James

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Re: [asterisk-users] Testing Asterisk with media - sipp

[Daniel - Asterisk]

Thank you Alex,

It's running without errors now and I can see the media flowing with
'rtp set debug on' but I can't still hear anything on the Asterisk's
peers, any advice?

Elder

2011/7/4, Alex Balashov :
> 488 means no mutually acceptable codecs were negotiated between the
> endpoints.
>
> --
> 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/
>
> On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk  wrote:
>
>> I'm trying to get working SIPp with media but something is wrong (it's
>> working well without media), please help:
>>
>> This is the command I send at SIPp server:
>>   ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
>>
>> This is the result I see:
>>   Last Error: Aborting call on unexpected message for Call-Id
>> '19-12768@12...
>>
>> What I see at sipp's logs:
>>
>> 2011-06-28  14:32:57:6241309289577.624809: Aborting call on
>> unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100'
>> (index 1), received 'SIP/2.0 488 Not acceptable here
>>
>> Via: SIP/2.0/UDP
>> 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253
>> From: sipp ;tag=12768SIPpTag091
>> To: sut ;tag=as3614adc3
>> Call-ID: 1-12768@127.0.0.1
>> CSeq: 1 INVITE
>> Server: Asterisk PBX 1.8.4.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Content-Length: 0
>>
>> This is my asterisk 1.8's configuration:
>>
>> sip.conf
>> [sipp]
>> type=friend
>> context=sipp
>> host=dynamic
>> port=6000
>> user=sipp
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>>
>> extensions.conf:
>> [sipp]
>> exten => 2005,1,Answer
>> same=>n,Dial(SIP/intern,30)
>> same=>n,Hangup()
>>
>> exten => 2006,1,Answer()
>> same=> n,WaitMusicOnHold(4)
>> same=> n,Hangup()
>>
>>
>> I'm using sipp.3.1.src.tar.gz and I have installed it this way:
>> ..sip.svn# make pcapplay
>>
>> Thanks in advance.
>>
>> Elder
>> --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Enviado desde mi dispositivo móvil

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Re: [asterisk-users] Agent Login, Logout, Ready, Not Ready from the CTI application

[Jim Dickenson]

You need to use the AMI interface an deal with the events that are give to you.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote:

> Hi All;
> 
> We know that agents can login and logout from the phone handset. But if we 
> need the login, logout, ready and not ready to be from an application and to 
> be integrated with the CRM, how to acheive this?
> 
> Normally in Cisco and AVAYA, they use CTI integration and the CTI client 
> (which is embded in the CRM application) will receive the the caller id or 
> information via that CTI client.
> 
> How this to be done in Asterisk?
> 
> By the way: is the ready and not ready in Asterisk?
> 
> Regards
> Bilal
> 
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[asterisk-users] Agent Login, Logout, Ready, Not Ready from the CTI application

[bilal ghayyad]

Hi All;

We know that agents can login and logout from the phone handset. But if we need 
the login, logout, ready and not ready to be from an application and to be 
integrated with the CRM, how to acheive this?

Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which 
is embded in the CRM application) will receive the the caller id or information 
via that CTI client.

How this to be done in Asterisk?

By the way: is the ready and not ready in Asterisk?

Regards
Bilal

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Re: [asterisk-users] Testing Asterisk with media - sipp

[Alex Balashov]

488 means no mutually acceptable codecs were negotiated between the endpoints.

--
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/

On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk  wrote:

> I'm trying to get working SIPp with media but something is wrong (it's 
> working well without media), please help:
> 
> This is the command I send at SIPp server: 
>   ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
> 
> This is the result I see:
>   Last Error: Aborting call on unexpected message for Call-Id 
> '19-12768@12...
> 
> What I see at sipp's logs:
> 
> 2011-06-28  14:32:57:6241309289577.624809: Aborting call on 
> unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' 
> (index 1), received 'SIP/2.0 488 Not acceptable here
> 
> Via: SIP/2.0/UDP 
> 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253
> From: sipp ;tag=12768SIPpTag091
> To: sut ;tag=as3614adc3
> Call-ID: 1-12768@127.0.0.1
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.4.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> 
> This is my asterisk 1.8's configuration:
> 
> sip.conf
> [sipp]
> type=friend
> context=sipp
> host=dynamic
> port=6000
> user=sipp
> canreinvite=no
> disallow=all
> allow=ulaw
> 
> extensions.conf:
> [sipp]
> exten => 2005,1,Answer
> same=>n,Dial(SIP/intern,30)
> same=>n,Hangup()
> 
> exten => 2006,1,Answer()
> same=> n,WaitMusicOnHold(4)
> same=> n,Hangup()
> 
> 
> I'm using sipp.3.1.src.tar.gz and I have installed it this way:
> ..sip.svn# make pcapplay
> 
> Thanks in advance.
> 
> Elder
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[asterisk-users] Testing Asterisk with media - sipp

[Daniel - Asterisk]

I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:

This is the command I send at SIPp server:
  ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err

This is the result I see:
  Last Error: Aborting call on unexpected message for Call-Id
'19-12768@12...

What I see at sipp's logs:

2011-06-28  14:32:57:6241309289577.624809: Aborting call on
unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100'
(index 1), received 'SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-12768-1-0;received=192.168.1.253
From: sipp ;tag=12768SIPpTag091
To: sut ;tag=as3614adc3
Call-ID: 1-12768@127.0.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

This is my asterisk 1.8's configuration:

*sip.conf*
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=ulaw
*
*
*extensions.conf:*
[sipp]
exten => 2005,1,Answer
same=>n,Dial(SIP/intern,30)
same=>n,Hangup()

exten => 2006,1,Answer()
same=> n,WaitMusicOnHold(4)
same=> n,Hangup()


I'm using sipp.3.1.src.tar.gz and I have installed it this way:
..sip.svn# make pcapplay

Thanks in advance.

Elder
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[asterisk-users] RINGNOANSWER events in queue log

[Alex Vishnev]

Does anyone know why i would get this RINGNOANSWER events in queue_log when 
clearly the agent is busy and call-waiting is disabled.

1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0
1309550632|1309550533.399961|2253|Local/11@from-internal/n|COMPLETECALLER|1|74|1
1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zz
1309550666|1309550640.399969|2253|Local/01@from-internal/n|CONNECT|3|1309550663.399971|0
//here it looks like Agent01 got the call.
1309550671|1309550648.399970|2525|NONE|ENTERQUEUE||zzz
1309550671|1309550648.399970|2525|Local/05@from-internal/n|RINGNOANSWER|0
1309550671|1309550648.399970|2525|Local/01@from-internal/n|RINGNOANSWER|0
// why is the system trying that channel for agent01 again?
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Re: [asterisk-users] stream rtp from asterisk

[Johan Wilfer]

On 2011-07-04 15:07, Marcus Kvarsell wrote:
> Sending the rtp-data to external server. One example which I have not gotten 
> to work is this below:
>
> http://oreka.sourceforge.net/
>
> September 02, 2009: Asterisk interception via Xorcom Asterisk patch
>
> Added support for recording of Asterisk voice calls (TDM and IP) using 
> Xorcoms Asterisk patch. See here.
>
> If there is any folk out there that has knowledge of this or any similar 
> software I would be very happy if you could help me get this to work.
>
> / Marcus
http://oreka.sourceforge.net/oreka-user-manual.html#gettingvoiptraffic

Seems they propose setting the switch in "mirror/monitoring-mode" and
sniff the traffic on another server.
Normal managed and smart-switches support this option... Or you can
install the software on the asterisk server.

/Johan
>
>
>
>
> -Ursprungligt meddelande-
> Från: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov
> Skickat: den 4 juli 2011 14:49
> Till: asterisk-users@lists.digium.com
> Ämne: Re: [asterisk-users] stream rtp from asterisk
>
> On 07/04/2011 06:58 AM, Marcus Kvarsell wrote:
>
>> Anybody familiar with streaming rtp from asterisk. Preferably with the 
>> xorcom asterisk patch which streams rtp from asterisk to oreka audio 
>> server. Any ideas will do just fine though!
> Can you clarify what you mean by "streaming"?
>
> --
> 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/
>
> --
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>
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-- 
Med vänlig hälsning

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JT Tech | Utvecklare webb: http://jttech.se
direkt: +46 31 380 91 01  support: +46 31 380 91 00


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Re: [asterisk-users] Mixmonitor concept's question

[virendra bhati]

Hi

Your suggestion is right if we want different recording for all channels.

But my problem is that I want to know if more user call the same conference
at different time gape(difference) then mixmonitor will take single asterisk
thread for recording or multiple thread for recoding.



On Mon, Jul 4, 2011 at 7:13 PM, Earl  wrote:

> On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote:
> > [RecordPrompts]
> >
> > exten => ,1,Answer()
> > exten => ,n,NoOp(WelCome to conference section)
> > exten => ,n,Playback(ConfDemoWC)
> > exten =>
> >
> ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w
> > av,ab) exten => ,n,Konference(${EXTEN},ADRSV)
> >
> > Hi
> >
> > My basic doubt is that if 1 or more  person call  extension then
> > recording will be started by asterisk Mixmonitor application.
> >
> > So basic question will come into mind that all calls will start recording
> > it means more then 1 thread will start by asterisk for only recording
> > purpose but finally 1 file will be save at disk.
> >
> > Am I right ?
> >
> >
> > -
> > Thanks and regards
> >
> >  Virendra Bhati
> > +91-9172341457
> > Software Engineer
>
> Hi Virendra,
>
> It has been my experience that if two or more calls happen in the same
> second
> (which is the least identifier you are using to make the file unique) then
> the
> result is that only one file is created. (the other(s) are not)
>
> To solve that issue, you can use ${UNIQUEID}, something like:
>
> exten => 6000,63,Set(CALLTIME=${STRFTIME(${EPOCH},,
> %C%y%m%d%H%M%S)}_${UNIQUEID})
> exten => 6000,64,Set(CALLFILENAME=6000-${CALLTIME})
> exten => 6000,65,Set(CMD='/opt/lame/l.sh '${CALLFILENAME})
> exten => 6000,66,MixMonitor(${CALLFILENAME}.wav|v(2)V(0)|${CMD})
>
> Note that the 1st exten line wraps here in the email but it should be all
> on
> one line.
>
> earl
>
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-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] Mixmonitor concept's question

[Earl]

On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote:
> [RecordPrompts]
> 
> exten => ,1,Answer()
> exten => ,n,NoOp(WelCome to conference section)
> exten => ,n,Playback(ConfDemoWC)
> exten =>
> ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w
> av,ab) exten => ,n,Konference(${EXTEN},ADRSV)
> 
> Hi
> 
> My basic doubt is that if 1 or more  person call  extension then
> recording will be started by asterisk Mixmonitor application.
> 
> So basic question will come into mind that all calls will start recording
> it means more then 1 thread will start by asterisk for only recording
> purpose but finally 1 file will be save at disk.
> 
> Am I right ?
> 
> 
> -
> Thanks and regards
> 
>  Virendra Bhati
> +91-9172341457
> Software Engineer

Hi Virendra,

It has been my experience that if two or more calls happen in the same second 
(which is the least identifier you are using to make the file unique) then the 
result is that only one file is created. (the other(s) are not)

To solve that issue, you can use ${UNIQUEID}, something like:

exten => 6000,63,Set(CALLTIME=${STRFTIME(${EPOCH},,
%C%y%m%d%H%M%S)}_${UNIQUEID})
exten => 6000,64,Set(CALLFILENAME=6000-${CALLTIME})
exten => 6000,65,Set(CMD='/opt/lame/l.sh '${CALLFILENAME})
exten => 6000,66,MixMonitor(${CALLFILENAME}.wav|v(2)V(0)|${CMD})

Note that the 1st exten line wraps here in the email but it should be all on 
one line.

earl

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Re: [asterisk-users] stream rtp from asterisk

[Marcus Kvarsell]

Sending the rtp-data to external server. One example which I have not gotten to 
work is this below:

http://oreka.sourceforge.net/

September 02, 2009: Asterisk interception via Xorcom Asterisk patch

Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms 
Asterisk patch. See here.

If there is any folk out there that has knowledge of this or any similar 
software I would be very happy if you could help me get this to work.

/ Marcus




-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov
Skickat: den 4 juli 2011 14:49
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] stream rtp from asterisk

On 07/04/2011 06:58 AM, Marcus Kvarsell wrote:

> Anybody familiar with streaming rtp from asterisk. Preferably with the 
> xorcom asterisk patch which streams rtp from asterisk to oreka audio 
> server. Any ideas will do just fine though!

Can you clarify what you mean by "streaming"?

--
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/

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Re: [asterisk-users] stream rtp from asterisk

[Alex Balashov]

On 07/04/2011 06:58 AM, Marcus Kvarsell wrote:


Anybody familiar with streaming rtp from asterisk. Preferably with the
xorcom asterisk patch which streams rtp from asterisk to oreka audio
server. Any ideas will do just fine though!


Can you clarify what you mean by "streaming"?

--
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/

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[asterisk-users] stream rtp from asterisk

[Marcus Kvarsell]

Hi!

Anybody familiar with streaming rtp from asterisk. Preferably with the
xorcom asterisk patch which streams rtp from asterisk to oreka audio
server. Any ideas will do just fine though!

Regards / Marcus


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Re: [asterisk-users] Distributing the incoming calls and the huntgroup

[bilal ghayyad]

Just to be sure that I am working in the right direction.

To do ACD, then I have to configure queue.conf and agent.conf?

One more question: if the agent needs to be in the NotReady state, then how 
this can be acheived?

Regards
Bilal

-
 
> FreeBPX calls them Ring Groups, you can look in to
> that.  Or you could
> use a small ACD group.
> 

> Hi All;
> 
> To be able to distribute the incoming calls on a group of
> extensions, is
> there huntgroup in Asterisk? Or what I have to use?
> 
> I need first call to be send for extension 500 and second
> call to be
> send for extension 501 and third call to be send for
> extension 502 and
> fourth call to be send again for extension 501 and so on ..
> 
> 
> I searched for huntgroup in Asterisk, but did not find any
> thing related
> to huntgroup in asterisk ! It look like there is not
> huntgroup in
> asterisk?!
> 
> So how to distribute the calls?
> 
> Regards
> Bilal
> 


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Re: [asterisk-users] how to set to make a call through a fixed ipon a 2 ips server?

[cnasterisk]

thanks, Faisal Hanif! i will try it!


2011-07-04 



cnasterisk 



发件人： Faisal Hanif 
发送时间： 2011-07-04  15:59:34 
收件人： 'Asterisk Users Mailing List - Non-Commercial Discussion' 
抄送： 
主题： Re: [asterisk-users] how to set to make a call through a fixed ipon a 2 ips 
server? 
 
Hi,
 
I don’t think there is a way for it inside asterisk but you achieve it by 
adding static route in Linux routing table and make interface having that IP as 
default route for the interested IPs traffic.
Regards,
Faisal
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cnasterisk
Sent: Monday, July 04, 2011 10:40 AM
To: asterisk-users
Subject: [asterisk-users] how to set to make a call through a fixed ip on a 2 
ips server?
 
Hi all,
I have a server runing asterisk 1.8, and the server has 2 different ip address
if i want to make a call from a  sip trunk with a fixed ip from the 2 ips, how 
to do?
 
 
 
2011-07-04 



cnasterisk 
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[asterisk-users] Mixmonitor concept's question

[virendra bhati]

[RecordPrompts]

exten => ,1,Answer()
exten => ,n,NoOp(WelCome to conference section)
exten => ,n,Playback(ConfDemoWC)
exten =>
,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.wav,ab)
exten => ,n,Konference(${EXTEN},ADRSV)

Hi

My basic doubt is that if 1 or more  person call  extension then
recording will be started by asterisk Mixmonitor application.

So basic question will come into mind that all calls will start recording it
means more then 1 thread will start by asterisk for only recording purpose
but finally 1 file will be save at disk.

Am I right ?


-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] SIP Peer Name Variable

[Faisal Hanif]

When you make a call asterisk always create a channel named as below,

 CheannelType/PeerName-uniquecode
 Like
 SIP/jon-312abf

So here jon is the peer name. This can help you to identify a peer as long
as A-Leg is active.

Regards,

Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Sunday, July 03, 2011 6:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Peer Name Variable



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan 
> Journo
> Sent: Saturday, July 02, 2011 8:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] SIP Peer Name Variable
>
> Hi,
>
>
>
> Is there a variable that contains the Sip Peer name?
>
> I was using ${CALLERID(num)} for outgoing calls, but when a call is 
> being transferred, that variable contains something else.
>
>
>
> I need a variable that is always set to the SIP Peer's name.

pbx*CLI> core show function CHANNEL

  -= Info about function 'CHANNEL' =-

[Synopsis]
Gets/sets various pieces of information about the channel.

[Description]
Gets/sets various pieces of information about the channel, additional 
may be available from the channel driver; see its documentation for details.
Any  requested that is not available on the current channel will
return an empty string.

[Syntax]
CHANNEL(item)

[Arguments]
item
Standard items (provided by all channel technologies) are:
audioreadformat - R/O format currently being read.
audionativeformat - R/O format used natively for audio.
audiowriteformat - R/O format currently being written.
callgroup - R/W call groups for call pickup.
channeltype - R/O technology used for channel.
checkhangup - R/O Whether the channel is hanging up (1/0)
language - R/W language for sounds played.
musicclass - R/W class (from musiconhold.conf) for hold music.
name - The name of the channel
parkinglot - R/W parkinglot for parking.
rxgain - R/W set rxgain level on channel drivers that support it.
secure_bridge_signaling - Whether or not channels bridged to this
channel require secure signaling
secure_bridge_media - Whether or not channels bridged to this channel
require secure media
state - R/O state for channel
tonezone - R/W zone for indications played
transfercapability - R/W ISDN Transfer Capability, one of:
SPEECH
DIGITAL
RESTRICTED_DIGITAL
3K1AUDIO
DIGITAL_W_TONES
VIDEO
txgain - R/W set txgain level on channel drivers that support it.
videonativeformat - R/O format used natively for video
trace - R/W whether or not context tracing is enabled, only available
*if CHANNEL_TRACE is defined*.
*chan_sip* provides the following additional options:
peerip - R/O Get the IP address of the peer.
recvip - R/O Get the source IP address of the peer.
from - R/O Get the URI from the From: header.
uri - R/O Get the URI from the Contact: header.
useragent - R/O Get the useragent.
peername - R/O Get the name of the peer.
t38passthrough - R/O '1' if T38 is offered or enabled in this channel,
otherwise '0'
rtpqos - R/O Get QOS information about the RTP stream
This option takes two additional arguments:
Argument 1:
 'audio' Get data about the audio stream
 'video' Get data about the video stream
 'text'  Get data about the text stream
Argument 2:
 'local_ssrc'Local SSRC (stream ID)
 'local_lostpackets' Local lost packets
 'local_jitter'  Local calculated jitter
 'local_maxjitter'   Local calculated jitter (maximum)
 'local_minjitter'   Local calculated jitter (minimum)
 'local_normdevjitter'Local calculated jitter (normal
 deviation)
 'local_stdevjitter' Local calculated jitter (standard
 deviation)
 'local_count'   Number of received packets
 'remote_ssrc'   Remote SSRC (stream ID)
 'remote_lostpackets'Remote lost packets
 'remote_jitter' Remote reported jitter
 'remote_maxjitter'  Remote calculated jitter (maximum)
 'remote_minjitter'  Remote calculated jitter (minimum)
 'remote_normdevjitter'Remote calculated jitter (normal
 deviation)
 'remote_stdevjitter'Remote calculated jitter (standard
 deviation)
 'remote_count'  Number of transmitted packets
 'rtt'   Round trip time
 'maxrtt'Round trip time (maximum)
 'minrtt'Round trip time (minimum)
 'normdevrtt'Round trip time (normal deviation)
 'stdevrtt'  Round trip time (standard deviation)
  

Re: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server?

[Faisal Hanif]

Hi,

 

I don't think there is a way for it inside asterisk but you achieve it by
adding static route in Linux routing table and make interface having that IP
as default route for the interested IPs traffic.

Regards,

Faisal

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cnasterisk
Sent: Monday, July 04, 2011 10:40 AM
To: asterisk-users
Subject: [asterisk-users] how to set to make a call through a fixed ip on a
2 ips server?

 

Hi all,

I have a server runing asterisk 1.8, and the server has 2 different ip
address

if i want to make a call from a  sip trunk with a fixed ip from the 2 ips,
how to do?

 

 

 

2011-07-04 

  _  

cnasterisk 

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