Re: [asterisk-users] Asterisk crashes at "meetme kick all"

[Deka, Rajib IN MAA SL]

Thanks a lot Patrick.

Regards
Rajib Deka
Siemens Ltd.

--

Message: 7
Date: Mon, 17 Feb 2014 10:22:02 +0100
From: Patrick Laimbock 
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk crashes at "meetme kick all"
Message-ID: <5301d4ba.3040...@laimbock.com>
Content-Type: text/plain; charset=windows-1252; format=flowed

On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:
> Dear Forum,
> I have encountered a similar issue as below in Asterisk 10.0.0. 
> Asterisk crashed while executing ?meetme kick all? CLI command from 
> manager interface. The link says the issue has been closed however I 
> am not able to identify in which release of asterisk this issue has been 
> fixed.
> Please help.
> _https://issues.asterisk.org/jira/browse/ASTERISK-15741_

AFAICT this issue has not been fixed due to inactivity. Note the "Suspended due 
to lack of activity" remark. Also the 1.6 version mentioned in the bugreport is 
EOL. Version 10.0.0 you mentioned is also EOL so any bugreport you file against 
version 10.0.0 will not be acted upon unless you can reproduce it with the 
latest Asterisk version 11.x.x or 12.x.x.

I recommend you upgrade to an Asterisk LTS (long term support) version like the 
latest 11.x.x (currently 11.7.0). For more information about Asterisk LTS 
versions go to:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can report 
it at the Asterisk issue tracker at:

https://issues.asterisk.org/jira/secure/Dashboard.jspa

Before filing a bug please read the information at:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug

--
Patrick




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[asterisk-users] Asterisk crashes at "meetme kick all"

[Deka, Rajib IN MAA SL]

Dear Forum,

I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk 
crashed while executing "meetme kick all" CLI command from manager interface. 
The link says the issue has been closed however I am not able to identify in 
which release of asterisk this issue has been fixed. Please help.

https://issues.asterisk.org/jira/browse/ASTERISK-15741


With best regards,
Rajib Deka
Siemens Ltd.


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[asterisk-users] Asterisk is delaying DTMF INFO in meetme

[Deka, Rajib IN MAA SL]

Hi List,

We have a major issue while broadcasting DTMF using meetme application. We are 
sending DTMF to asterisk using SIP INFO message with duration 160.

INFO sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx:5060
From: ;tag=43
To: ;tag=9753.0207
Call-ID: xxx@xxx
CSeq: 25634 INFO
Content-Length: 26
Content-Type: application/dtmf-relay
Signal= 2\r\n
Duration= 160\r\n

[Nov 19 15:30:43] [1;32mDEBUG[0m[2966]:[1;37mchan_sip.c[0m:[1;37m24896[0m 
[1;37mhandle_incoming[0m:  Received INFO (13) - Command in SIP INFO
Receiving INFO!
* DTMF-relay event received: 2
[KCentos-2*CLI> [0K[Nov 19 15:30:43] [[1;32mDTMF[[0m[17988]: 
[[1;37mchannel.c[[0m:[[1;37m3978[[0m [[1;37m__ast_read[[0m: DTMF end '2' 
received on SIP/16222-0037, duration 160 ms
M[[KCentos-2*CLI> M[[0K[Nov 19 15:30:43] [[1;32mDTMF[[0m[17988]: 
[[1;37mchannel.c[[0m:[[1;37m4004[[0m [[1;37m__ast_read[[0m: DTMF begin 
emulation of '2' with duration 160 queued on SIP/16222-0037
[Nov 19 15:30:45] [[1;32mDTMF[[0m[17988]: [[1;37mchannel.c[[0m:[[1;37m4096[[0m 
[[1;37m__ast_read[[0m: DTMF end emulation of '2' queued on SIP/16222-0037
[Nov 19 15:30:45] [[1;32mDEBUG[[0m[17988]: 
[[1;37mchan_sip.c[[0m:[[1;37m3328[[0m [[1;37m__sip_xmit[[0m: Trying to put 
'INFO sip:18' onto UDP socket destined for 132.186.230.236:6372


>From the above log  (Nov 19 15:30:43 and Nov 19 15:30:45)I can see that after 
>receiving SIP INFO asterisk is trying to regenerate the DTMF tone based on the 
>duration specified by the client. Which is OK, but latency observed in this 
>operation is more than 2 Sec in some cases and also asterisk changes the 
>duration field in SIP INFO message body. Please help us out to overcome this 
>problem as more than 2 sec latency is not acceptable in real-time scenarios. 
>Also if possible let us know (technically), whether it is a know issue in 
>asterisk.

Regards
Rajib
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[asterisk-users] MCID

[Deka, Rajib IN MAA SL]

Hello Forum,

Does any version of asterisk supports Malicious Communication Identification 
(MCID) using IP standard 3GPP TS 24.616? If yes how can I enable or configure 
it?

Regards
Rajib
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Re: [asterisk-users] Asterisk crash

[Deka, Rajib IN MAA SL]

Yes we can reproduce this crash scenario by running calls between portsip and 
Xlite soft phones. The issue we have observed is CODEC translation between iLBC 
and alaw with following warning messages,

[Sep  2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel 
SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 
(ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation 
path from 0x4 (ulaw) to 0x400 (ilbc)
[Sep  2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, 
while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel 
SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 
(ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation 
path from 0x4 (ulaw) to 0x400 (ilbc)
[Sep  2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, 
while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel 
SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 
(ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation 
path from 0x4 (ulaw) to 0x400 (ilbc)
[Sep  2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, 
while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
[Sep  2 15:59:53] WARNING[24418] channel.c: No path to translate from 
SIP/18252-0002d010 to SIP/18203-0002d01e
[Sep  2 15:59:53] WARNING[24418] channel.c: Can't make SIP/18252-0002d010 and 
SIP/18203-0002d01e compatible
[Sep  2 15:59:53] WARNING[24418] features.c: Bridge failed on channels 
SIP/18252-0002d010 and SIP/18203-0002d01e

We can reproduce the problem as below,
1. Call between Xlite(iLBC) to portsip(G711), RTP through asterisk.
2. portsip attended transfer the call to another portsip client
3. on complete transfer asterisk crashes (then started by safe_asterisk) with 
above warning.

FYI, we have not installed asterisk with iLBC support.

We will try to upgrade asterisk and try to reproduce this scenario.

Regards
Rajib

--

Message: 13
Date: Wed, 4 Sep 2013 09:28:12 -0500
From: Rusty Newton 
Subject: Re: [asterisk-users] Asterisk crash
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset=UTF-8

On Tue, Sep 3, 2013 at 4:17 AM, Deka, Rajib IN MAA SL
 wrote:

> In our lab asterisk has crashed due to some unknown reason and it has been
> restarted by safe_asterisk service. But before crash we can see lots of
> below log entry (asterisk version 1.8.9.3).

That is quite old. Lots of bugs (and several security issues) have
been fixed since then. Try the latest in the 1.8 branch.

For the crash , follow the instructions here

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

 and gather a backtrace after recompiling with the required options.
(preferably after upgrading to the latest 1.8, as there may have been
improvmen

> Sep  3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error
> of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported
> by protocol
>
> chan_sip.c: Purely numeric hostname, and not a peer--rejecting!

These messages alone don't show the whole picture.

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Collect a log with VERBOSE and DEBUG turned up to level 5, SIP debug
turned on, and pastebin that.

I'd wait until after you test with the latest in 1.8

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] Asterisk crash

[Deka, Rajib IN MAA SL]

Hello List,

In our lab asterisk has crashed due to some unknown reason and it has been 
restarted by safe_asterisk service. But before crash we can see lots of below 
log entry (asterisk version 1.8.9.3).

Sep  3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of 
packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by 
protocol
chan_sip.c: Purely numeric hostname, and not a peer--rejecting!

Did someone encounter this problem before? Please let me know.

Regards
Rajib
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[asterisk-users] SIP timers

[Deka, Rajib IN MAA SL]

Hello List,

I tried to change the following parameters in sip.conf file, but looks like it 
cannot be changed,

Defaut values:
;t1min=100

;timert1=500

;timerb=32000



I have changed to:
;t1min=100

timert1=100

timerb=6400

Sometime I can see too many retransmission of BYE to some of the UAs if UA is 
unreachable. Is there a way  that I can reduce the number of retransmission of 
BYE message?

Regards
Rajib
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Re: [asterisk-users] announcement to be played for attended

[Deka, Rajib IN MAA SL]

Thanks a lot Dona and jg for your inputs.

I'll try to find some way to do this from Dialplan or AMI and let you guys know 
soon. Please share if you have some more ideas.

Regards,
Rajib

Date: Tue, 11 Jun 2013 18:34:46 +0200
From: jg 
Subject: Re: [asterisk-users] announcement to be played for attended
transfer call
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <51b751a6.5000...@jgoettgens.de>
Content-Type: text/plain; charset=UTF-8; format=flowed

So, B transfers the call and after bridging to C, B should get an
announcement.

This is just an idea:
See whether you can dispatch the termination of the call leg B-C by
evaluating the DIALSTATUS variable. I am not sure whether you can see
this inside the dialplan, but you should get the event via AMI. This is
only the 1st part of the solution.

A general solution would require a lot of things or may not be possible
at all as you can transfer calls not only via Asterisk using DTMF
signalling, but also the SIP phones themselves might be capable of
transferring calls, thereby circumventing Asterisk.

jg

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[asterisk-users] announcement to be played for attended transfer call

[Deka, Rajib IN MAA SL]

Hello List,



I want to play an announcement for attended transfer calls. For example, "A" 
calls "B", "B" answers the call and transfers (attended) to "C" - once transfer 
is complete "B" should hear an announcement saying "you call has been 
transferred". Is there any configuration in asterisk to implement this behavior?



I have not used asterisk Transfer Dialplan application or feature.conf for 
configuring the transfer; however I am using SIP REFER from UA to request the 
transfer.



Regards,

Rajib





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[asterisk-users] asterisk MCID detection

[Deka, Rajib IN MAA SL]

Hello List,

Does asterisk 1.8 or 10 provides any events or Dialplan application to detect 
MCID (Malicious Caller Identification) in an incoming call?
Please provide any sample Dialplan if possible.

Thanks & Regards,
Rajib

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[asterisk-users] Reffered By header is missing from SIP INVITE in call transfer scenarios

[Deka, Rajib IN MAA SL]

Hi,
We are facing an issue with asterisk in the case of call-Transfer scenarios.
Our requirement is to identify whether an incoming call is a fresh incoming 
call or a Transferred call from some other clients.

We have a setup, where in the asterisk1.6 (as SIP server) is running in Linux 
machine, and three SIP clients(say A,B,C) registered to asterisk server are 
running in three different windows machines.

With the above said setup, there is a call made from SIP client-A to SIP 
client-B through asterisk. The incoming call got answered in SIP client-B and 
transferred the call to SIP client-C via asterisk.
Here the SIP client-B sends a REFER SIP message to Asterisk and a new INVITE 
(corresponds to the REFER SIP) is sent to SIP client-C. But there is no 
REFFERED BY Header added in the INVITE SIP message which is sent to SIP 
client-C.
Due to this we are not able to identify the incoming call as Transferred call.

So, we have two questions:
1)   Are there any configuration changes in Asterisk to solve this (so that 
the asterisk handles the transfer in SIP signaling)?
2)   Is there any other way in which we can identify a call as forwarded 
call?


Best regards,
Rajib
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[asterisk-users] Rgarding asterisk 10 stable release

[Deka, Rajib IN MAA SL]

Hello List,

We are eagerly waiting for stable release of Asterisk 10 as it support most 
awaited out of call messaging.
Can somebody please let me know when the stable release will be available for 
download?

Regards
Rajib Deka
Siemens Ltd.
India

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Re: [asterisk-users] Asterisk is delaying DTMF (SIP INFO) relay in MeetMe

[Deka, Rajib IN MAA SL]

Hello List,

I have seen that when ever asterisk gets a SIP INFO request from a SIP channel 
it generates the requested DTMF tone and writes to the destination channel also 
it forwards the SIP INFO message. As I am very new to this domain, it is really 
confusing me. Why not asterisk writes only the tone and can avoid forwarding of 
SIP INFO? I know I may be wrongly interpreted the things, Can somebody please 
explain me the scenario, if possible?

Thanks
Rajib


From: Deka, Rajib IN MAA SL
Sent: Monday, August 29, 2011 3:34 PM
To: 'asterisk-users@lists.digium.com'
Subject: Asterisk is delaying DTMF (SIP INFO) relay in MeetMe

Hello List,

We are using 'F' parameter in meetme Dialplan application to broadcast SIP INFO 
(1 and 0) as DTMF tone to all the participants.
The DTMF configuration for all the connected SIP clients is SIP INFO.

The problem we are seeing, asterisk is taking some time to broadcast the SIP 
INFO message to all the participants from the time of its appearance. The time 
latency varies from 1.5 sec to 6 sec. We have activated the highest debug and 
verbose level but we are not able to track down the problem. Please help us out 
to overcome this problem as 6 sec latency is not acceptable in real-time 
scenarios. Also if possible let us know (technically), whether it is a know 
issue in asterisk.

Regards,
Rajib
Siemens Ltd.
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[asterisk-users] Asterisk is delaying DTMF (SIP INFO) relay in MeetMe

[Deka, Rajib IN MAA SL]

Hello List,

We are using 'F' parameter in meetme Dialplan application to broadcast SIP INFO 
(1 and 0) as DTMF tone to all the participants.
The DTMF configuration for all the connected SIP clients is SIP INFO.

The problem we are seeing, asterisk is taking some time to broadcast the SIP 
INFO message to all the participants from the time of its appearance. The time 
latency varies from 1.5 sec to 6 sec. We have activated the highest debug and 
verbose level but we are not able to track down the problem. Please help us out 
to overcome this problem as 6 sec latency is not acceptable in real-time 
scenarios. Also if possible let us know (technically), whether it is a know 
issue in asterisk.

Regards,
Rajib
Siemens Ltd.
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[asterisk-users] AMI action PlayDTMF and SIP:INFO

[Deka, Rajib IN MAA SL]

Hello List,

Is it possible to send SIP:INFO to an active SIP channel using AMI action 
PlayDTMF?
I tried to send a DTMF digit 1 to my SIP client, but I did not get any SIP:INFO 
on wireshark trace.
In my sip.conf file dtmfmode=info.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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[asterisk-users] dialout time configuration

[Deka, Rajib IN MAA SL]

Hi List,

Is it possible to configure an infinite ring timeout for queue in asterisk?
I mean, the caller should be able to be in queue until and unless he 
disconnects the call.

Thanks,
Rajib


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Re: [asterisk-users] SIP Presence not working

[Deka, Rajib IN MAA SL]

Hi All,

Following message I got in console for an extension,

[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c:
<--- SIP read from UDP:132.186.230.70:7510 --->
SUBSCRIBE sip:18...@sip1.test.in SIP/2.0^M
Via: SIP/2.0/UDP 
132.186.230.70:7510;branch=z9hG4bK-d8754z-2b3b65532b3b6553-1---d8754z-;rport^M
Max-Forwards: 70^M
Contact: ^M
To: ^M
From: "18238";tag=2b3b6553^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 1 SUBSCRIBE^M
Subject: Available^M
Expires: 3600^M
Accept: multipart/related, application/rlmi+xml, application/pidf+xml^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, 
SUBSCRIBE, INFO^M
Supported: replaces^M
User-Agent: ^M
Event: presence^M
Content-Length: 0^M
^M

<->
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: --- (16 headers 0 lines) ---
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Ignoring this SUBSCRIBE request
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Found peer '18238' for '18238' from 
132.186.230.70:7510
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Looking for 18227 in 
test-local-outgoing (domain sip1.test.in)
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Scheduling destruction of SIP 
dialog 'MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.' in 361 ms (Method: 
SUBSCRIBE)
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c:
<--- Transmitting (no NAT) to 132.186.230.70:7510 --->
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
132.186.230.70:7510;branch=z9hG4bK-d8754z-2b3b65532b3b6553-1---d8754z-;received=132.186.230.70;rport=7510^M
From: "18238";tag=2b3b6553^M
To: ;tag=as6c37f730^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 1 SUBSCRIBE^M
Server: Asterisk PBX 1.6.2.16.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M
Supported: replaces, timer^M
Expires: 3600^M
Contact: ;expires=3600^M
Content-Length: 0^M

<>
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: set_destination: Parsing 
 for address/port to send to
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: set_destination: set destination to 
132.186.230.70, port 7510
[Jul  5 12:10:56] VERBOSE[3729] chan_sip.c: Reliably Transmitting (no NAT) to 
132.186.230.70:7510:
NOTIFY sip:18238@132.186.230.70:7510 SIP/2.0^M
Via: SIP/2.0/UDP 10.20.20.52:5060;branch=z9hG4bK0f6d2ae2;rport^M
Max-Forwards: 70^M
From: ;tag=as6c37f730^M
To: "18238";tag=2b3b6553^M
Contact: ^M
Call-ID: MmQ3ZDZiNzY4Y2FmNmU1OWU5NjUxODY5NTAyNTU3MDc.^M
CSeq: 119 NOTIFY^M
User-Agent: Asterisk PBX 1.6.2.16.1^M
Event: presence^M
Content-Type: application/pidf+xml^M
Subscription-State: active^M
Content-Length: 533^M
^M





Not online

sip:18...@sip1.siemens.in
closed





From: Deka, Rajib IN MAA SL
Sent: Tuesday, July 05, 2011 12:15 PM
To: 'asterisk-users@lists.digium.com'
Subject: SIP Presence not working

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] 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] Asterisk changing SIP INFO dtmf duration

[Deka, Rajib IN MAA SL]

Hello List,

We are facing a problem in broadcasting DTMF from MeetMe.
Our client is sending DTMF with duration=160 (in SIP INFO) to asterisk but 
asterisk is changing this header to different values like 162, 175 etc while 
broadcasting to all the participants. Is it possible to restrict asterisk from 
changing this header value or this is a common behavior of all the PBXs.

Regards,
Rajib


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Re: [asterisk-users] asterisk queue 'ringall' stratagy

[Deka, Rajib IN MAA SL]

Thanks a lot for all your comments.
Finally I have figured out the problem by looking into source code.
If callcounter=yes and notification is enabled for ringing or hold in sip.conf 
file, asterisk queue will not fork the new incoming call to the members already 
in ringing or inuse state.

Regards
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com


rajib,

You can use DIALGROUP function as well

On Mon, Jun 13, 2011 at 7:36 PM, Mike  wrote:

> Quite simply: don?t use a queue.  Simply ring all phones at the same time
> using Dial(SIP/phone1&SIP/phone2&?.)
>
>
>
> A queue will only send the first call until it is answered, then move on to
> the second one (I may be simplifying a bit)
>
>
>
> Mike
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Deka, Rajib IN MAA
> SL
> *Sent:* Monday, June 13, 2011 6:44 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] asterisk queue 'ringall' stratagy
>
>
>
> Hi List,
>
>
>
> I have faced a problem in asterisk queue implementation.
>
>
>
> I configured a queue with ?ringall? strategy and ?ringinuse=yes? in
> queues.conf. If three calls come to this queue in parallel, the logged in
> queue agent used to get only one call (may be the first one), not all the
> calls waiting in the queue at a time. Once the agent answers the call the
> next call is displayed.
>
> I want to display all the waiting calls on the agent?s desktop. Is it
> possible to do, if yes how? Please help me with this.
>
>
>
> Regards,
>
> Rajib
>
>
>
> *Rajib Deka*
>
> SIEMENS Ltd.
>
> Robert V Chandran Tower, First Floor, West Wing,
>
> #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
>
> www.siemens.com
>
>
>
> Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
>
>
>
>
> --
>
> Important notice: This e-mail and any attachment there to contains
> corporate proprietary information. If you have received it by mistake,
> please notify us immediately by reply e-mail and delete this e-mail and its
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[asterisk-users] asterisk queue 'ringall' stratagy

[Deka, Rajib IN MAA SL]

Hi List,

I have faced a problem in asterisk queue implementation.

I configured a queue with 'ringall' strategy and 'ringinuse=yes' in 
queues.conf. If three calls come to this queue in parallel, the logged in queue 
agent used to get only one call (may be the first one), not all the calls 
waiting in the queue at a time. Once the agent answers the call the next call 
is displayed.
I want to display all the waiting calls on the agent's desktop. Is it possible 
to do, if yes how? Please help me with this.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] DAHDi installation problem

[Deka, Rajib IN MAA SL]

Thank you very much Ruffell and Patrick.
The problem was basic. The OS was missing with correct kernel headers.
We installed correct kernel headers and its working fine now.

Regards,
Rajib

On Mon, May 30, 2011 at 02:29:37PM +0200, Patrick Lists wrote:
> On 05/30/2011 10:03 AM, Deka, Rajib IN MAA SL wrote:
>> Hello List,
>>
>> What version of DAHDi should be installed for CentOS Kernel version
>> 2.16.18?194.el5.
>
> I would use the latest DAHDI version which is currently:
>
> DAHDI tools: 2.4.1
> DAHDI linux: 2.4.1.2
>
> You can find them at http://www.asterisk.org/downloads
>
>> We do not have access to yum in our network, so we need to install a
>> specific version with respect to kernel version.
>
> Not being able to use yum does make things a bit challenging.
>
>> Or, what update to be downloaded and applied to CentOS kernel to install
>> a specific version of DAHDi.
>
> DAHDI should work fine with your current kernel and it should work fine
> with the latest kernel.
>
> The kernel version you mentioned suggests that you are using CentOS 5.5
> or RHEL 5.5. I would get the 5.6 DVD iso and upgrade the box to 5.6 and
> then download (to a USB stick) and install (from the USB stick) all the
> updates that are available since the release of 5.6.

Hi Rajib. In addition to what Patrick wrote I would ask that you write
what the error messages that you're seeing when you try to install. It
could be something as basic as you don't have the correct kernel headers
installed.

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org



--


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[asterisk-users] DAHDi installation problem

[Deka, Rajib IN MAA SL]

Hello List,

What version of DAHDi should be installed for CentOS Kernel version 
2.16.18-194.el5.
We do not have access to yum in our network, so we need to install a specific 
version with respect to kernel version.

Or, what update to be downloaded and applied to CentOS kernel to install a 
specific version of DAHDi.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] asterisk hint SIP presence

[Deka, Rajib IN MAA SL]

Thank you John!
I too figure out the way using 'sip show subscription'.

Regards,
Rajib


> Hello List,

> Asterisk CLI command ?core show hints? gives the list of hint extension
> configured and its presence status.
>
> In command output there is a field called ?watchers? and it contains a
> numeric value of number of subscriptions? registered for that particular
> extension.
>
> So, is there any CLI command to check who the watchers for an extension are?
>

I use 'sip show subscriptions' to see what peers watch which hint.


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Re: [asterisk-users] asterisk HA for queue calls

[Deka, Rajib IN MAA SL]

Hi Dhaval,

Thanks for your much appreciable reply.
Sorry for late reply as I was out of office.
We considered the situation that pending queue call cannot be retrieved during 
failover, and hence it's ok with us if we loose the calls also.

Regards,
Rajib

Date: Wed, 4 May 2011 14:15:59 +0530
From: DHAVAL INDRODIYA 
Subject: Re: [asterisk-users] asterisk HA for queue calls
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

Hi Rajib,

I think It is not possible with asterisk , as primary server goes down it
will stop asterisk services so once asterisk service down i think all
connected calls to queue will hangup automatically, and you cannot retrive
those calls as they all are disconnected .

I think you need to consider more on load balancing per asterisk server in
that case the problem of Availability is solved to some level, If You using
SIP protocol then you can think of OPENSER and from that you can use
loadbalancer which routed calls in a way an depend on machine strength.

I hope this idea will useful to solve your requirement.

Regards
Dhaval

On Wed, May 4, 2011 at 1:13 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hello List,
>
>
>
> We are running two asterisk machines in virtual IP as primary and secondary
> server.
>
> Initially virtual IP will be active in primary server; during the failure
> of primary secondary will get the virtual IP.
>
>
>
> Is there any way to retrieve pending queue calls from primary to secondary,
> in case primary fails?
>
> Does asterisk provide any interface to do it or we have to write some
> application on asterisk to do the same.
>
>
>
> Regards,
>
> Rajib
>
>
>

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Re: [asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

It's working now. I removed the 'm' flag from the meetme Dialplan. It was my 
mistake. Asterisk is working fine.
Exten => 100,1,MeetMe(100,dF)

Regards,
Rajib

____
From: Deka, Rajib IN MAA SL
Sent: Wednesday, May 11, 2011 5:35 PM
To: 'asterisk-users@lists.digium.com'
Subject: no audio with SIP:INFO in meetme

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com<http://www.siemens.com>

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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[asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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[asterisk-users] asterisk HA for queue calls

[Deka, Rajib IN MAA SL]

Hello List,

We are running two asterisk machines in virtual IP as primary and secondary 
server.
Initially virtual IP will be active in primary server; during the failure of 
primary secondary will get the virtual IP.

Is there any way to retrieve pending queue calls from primary to secondary, in 
case primary fails?
Does asterisk provide any interface to do it or we have to write some 
application on asterisk to do the same.

Regards,
Rajib



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[asterisk-users] play audio file to destination SIP channel on attended call transfer

[Deka, Rajib IN MAA SL]

Hello List,

Please help with the following problem,

I have a situation, where I need to play an audio announcement to the caller 
SIP channel once an attended transfer is successful. The attended transfer is 
done from client. I can see a transfer event in AMI. I am not using 'T/t' 
option in dial() command. The transfer is completely on client side using SIP 
signaling.
1. A calls B
2. B answers A
3. A calls C
4. B is in hold
5. C answers
6. B transfer A->C
7. Play an announcement to A 'your call transferred'
8. A and C bridged.

I have enabled "xfersound = " in features.conf.
But it's not working once I transfer the call.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND

[Deka, Rajib IN MAA SL]

Thanks a lot guys.
We planning to use different approach like maintaining a separate extension for 
sending and receiving in dialog SIP:MESSAGE using SendText and RECEIVE TEXT.

Regards,
Rajib

DHAVAL INDRODIYA  wrote:
>
> is your problem solved or not

It will take a lot more time than that to try out the suggestions!

> On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL <
> rajib.d...@siemens.com> wrote:
>
> > Thanks a lot Tony and Dhaval for your much appreciable suggestions.
> >
> > Regards,
> > Rajib
> >
> > Rajib Deka
> > SIEMENS Ltd.
> > Robert V Chandran Tower, First Floor, West Wing,
> > #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
> > www.siemens.com
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org


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Re: [asterisk-users] No voice in MeetMe for SIP with

[Deka, Rajib IN MAA SL]

Thanks a lot Tony and Dhaval for your much appreciable suggestions.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com

Date: Wed, 20 Apr 2011 13:55:25 +0530
From: DHAVAL INDRODIYA 
Subject: Re: [asterisk-users] No voice in MeetMe for SIP with
AGI_BACKGROUND
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

hey try with app_rpt in asterisk

regards
dhaval

On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield wrote:

> In article <
> 2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net
> >,
> Deka, Rajib IN MAA SL  wrote:
> >
> > The requirement is little complicated as it is H/W specific.
> > Basically we are integrating a radio gateway (SIP) with asterisk. The
> gateway will be
> > connected to a meetme room, so that any operator (with IP phone
> registered as SIP user to
> > asterisk) can login to the room and listen to radio communications and
> talk.
> >
> > Using a PTT button someone can talk on a radio channel. Once someone
> presses the PTT button
> > a SIP MESSAGE is sent to the gateway with a string as payload to enable
> half duplex
> > communication. So, we were planning to run an AGI script with meetme
> (AGI_BACKGROUND) to
> > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and
> to generate a
> > VarSet AMI event.
> >
> > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE
> -> radio gateway
> > And vise versa.
> >
> > Any suggestions on the above scenario.
>
> I don't think it can be done without making modifications to Asterisk.
>
> The first thing I would do, if you haven't done so already, would be to
> try it without MeetMe:
>
> Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE ->
> radio gateway
>
> If that works, then it would suggest that the SIP MESSAGE is
> successfully getting translated into an ast_frame, which is then getting
> translated back into a SIP MESSAGE. If that is not happening, you might
> need to add some code to chan_sip.c to do those steps.
>
> Once Asterisk is converting the message to and from an ast_frame, the
> next step would be to add some code to app_meetme.c in the conf_run()
> function, to pass those frames through, in the same way as DTMF frames
> get passed through when the F option is enabled.
>
> Presumably the messages represent PTT PRESS and PTT RELEASE. You will
> need to decide what to do if you have two operators connected and they
> both press the PTT.
>
> You might also need to automatically unmute or mute the operator
> channel when their PTT is pressed or released. That could also be done
> within the MeetMe code.
>
> There may be other approaches too...
>
> Hope this helps!
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
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Re: [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND

[Deka, Rajib IN MAA SL]

Hello List,

The requirement is little complicated as it is H/W specific.
Basically we are integrating a radio gateway (SIP) with asterisk. The gateway 
will be connected to a meetme room, so that any operator (with IP phone 
registered as SIP user to asterisk) can login to the room and listen to radio 
communications and talk.

Using a PTT button someone can talk on a radio channel. Once someone presses 
the PTT button a SIP MESSAGE is sent to the gateway with a string as payload to 
enable half duplex communication. So, we were planning to run an AGI script 
with meetme (AGI_BACKGROUND) to receive the MESSAGE (using AGI command 'RECEIVE 
TEXT') from both ends and to generate a VarSet AMI event.

Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE -> 
radio gateway
And vise versa.

Any suggestions on the above scenario.

Regards,
Rajib

Date: Tue, 19 Apr 2011 10:40:05 + (UTC)
From: t...@softins.co.uk (Tony Mountifield)
Subject: Re: [asterisk-users] No voice in MeetMe for SIP with
AGI_BACKGROUND
To: asterisk-users@lists.digium.com
Message-ID: 

In article 
<2658e54b540d284981ea57e6a549ea70abd1f7c...@inblrk77m1msx.in002.siemens.net>,
Deka, Rajib IN MAA SL  wrote:
>
> I have seen from the following link that, for SIP channels there is no audio 
> communication
> possible in MeetMe with AGI_BACKGROUND.
> http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
>
> Currently we are using asterisk-1.6.2 and the problem still persists. Is 
> there any solution
> available to overcome this problem? According to our requirement, we have to 
> run an AGI
> script in MeetMe.

The fact that background AGI in meetme only works with Zap channels
is a consequence of the original design of Meetme. See these two old
posts:

http://lists.digium.com/pipermail/asterisk-users/2004-August/050725.html
http://lists.digium.com/pipermail/asterisk-users/2004-August/050776.html

You will need to change to a different approach to solve your requirement.
Could you explain your original requirement? Then people on this list may
be able to suggest an alternative way to do it.

Cheers
Tony

--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org




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[asterisk-users] ConfBridge and AGI

[Deka, Rajib IN MAA SL]

Hello List,

Is it possible to run an AGI script in backgroung for all the associated SIP 
channels in ConfBridge Application? If yes how?
This can be done using 'b' parameter in MeetMe for non SIP channels.

Regards,
Rajib

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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[asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND

[Deka, Rajib IN MAA SL]

Hello List,

I have seen from the following link that, for SIP channels there is no audio 
communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe

Currently we are using asterisk-1.6.2 and the problem still persists. Is there 
any solution available to overcome this problem? According to our requirement, 
we have to run an AGI script in MeetMe.

Kind Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 21

[Deka, Rajib IN MAA SL]

Thank you Paul.
I have downloaded the code.

How out-of-call messaging can be configured in the Dialplan?

Regards,

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
Date: Thu, 07 Apr 2011 10:14:37 -0400
From: Paul Belanger 
Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support
To: asterisk-users@lists.digium.com
Message-ID: <4d9dc6cd.1060...@digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote:
>
> Is the following trunk has development version of out-of-call messaging 
> capability, also what is the version of asterisk,
> http://svn.asterisk.org/svn/asterisk/trunk/
>
I don't believe the branches has been merged into trunk, you can use
russellb's branch [1].

[1] http://svn.digium.com/svn/asterisk/team/russell/messaging/

--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] asterisk SIP MESSAGE method support

[Deka, Rajib IN MAA SL]

Is the following trunk has development version of out-of-call messaging 
capability, also what is the version of asterisk,
http://svn.asterisk.org/svn/asterisk/trunk/

Regards,
Rajib

--

Message: 10
Date: Thu, 7 Apr 2011 14:42:35 +0100
From: Steven Howes 
Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: 
Content-Type: text/plain; charset=us-ascii

On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:
> Is the following is the link for getting the source,
> http://svn.asterisk.org/svn/asterisk/trunk/

Please try not to reply to the entire digest..

S


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Re: [asterisk-users] asterisk SIP MESSAGE method support

[Deka, Rajib IN MAA SL]

Is the following is the link for getting the source,
http://svn.asterisk.org/svn/asterisk/trunk/

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
asterisk-users-requ...@lists.digium.com
Sent: Thursday, April 07, 2011 6:20 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 81, Issue 19

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

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When replying, please edit your Subject line so it is more specific
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Today's Topics:

   1. asterisk SIP MESSAGE method support (Deka, Rajib IN MAA SL)
   2. Re: Iptables configuration to handle brute force
  registrations? (Gilles)
   3. Re: BRI Configuration help me (mahesh katta)
   4. Re: Iptables configuration to handle brute,   force
  registrations? (Gilles)
   5. Compiling asterisk using NDK build (Nikhil)
   6. Re: asterisk SIP MESSAGE method support (Olivier)
   7. Re: BRI Configuration help me (Tzafrir Cohen)
   8. Re: Compiling asterisk using NDK build (Tzafrir Cohen)
   9. Re: BRI Configuration help me (mahesh katta)
  10. Re: Trunk form asterisk1 to asterisk2 fails (GiGi)
  11. Re: Asterisk 1.8.3 (Satish Patel)
  12. Re: Asterisk 1.8.3 (Bryant Zimmerman)
  13. Re: BRI Configuration help me (mahesh katta)


--

Message: 1
Date: Thu, 7 Apr 2011 14:54:23 +0530
From: "Deka, Rajib IN MAA SL" 
Subject: [asterisk-users] asterisk SIP MESSAGE method support
To: "asterisk-users@lists.digium.com"

Message-ID:

<2658e54b540d284981ea57e6a549ea70a592f02...@inblrk77m1msx.in002.siemens.net>

Content-Type: text/plain; charset="us-ascii"

Hello List,

I have found that asterisk supports only forwards in-dialog MESSAGE method. 
That is, if the MESSAGE method is sent within an active call.

But according our requirement we need to send MESSAGE method to the other leg 
without being in a call (general stateless proxy forward). Is it possible to do 
this in asterisk using some tricks?

Regards,

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com<http://www.siemens.com>

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Message: 2
Date: Thu, 07 Apr 2011 12:51:48 +0200
From: Gilles 
Subject: Re: [asterisk-users] Iptables configuration to handle brute
force   registrations?
To: asterisk-users@lists.digium.com
Message-ID: 
Content-Type: text/plain; charset=us-ascii

On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
 wrote:
>Have a look at these:

Thanks much Gordon. I'll study the scripts you mentionned. It looks
like iptables is good enough and I won't have to install a second tool
to watch the logs and reconfigure iptables on the fly.




--

Message: 3
Date: Thu, 7 Apr 2011 16:48:13 +0530
From: mahesh katta 
Subject: Re: [asterisk-users] BRI Configuration help me
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

Sir,

my files are in fistmail that is my configuration.

and till its disconnecting the line



On Thu, Apr 7, 2011 at 2:35 PM, Tzafrir Cohen wrote:

> Hi,
>
> Un-top-posting
>
> On Thu, Apr 07, 2011 at 12:53:18PM +0530, mahesh katta wrote:
> >
> > On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen  >wrote:
> >
> > > On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
> > > > Sir,
> > > >
> > > > i am using goautodial server , bri card is showing ok but when i try
> to
> > > call
> > > > that showing below ,
>

[asterisk-users] asterisk SIP MESSAGE method support

[Deka, Rajib IN MAA SL]

Hello List,

I have found that asterisk supports only forwards in-dialog MESSAGE method. 
That is, if the MESSAGE method is sent within an active call.

But according our requirement we need to send MESSAGE method to the other leg 
without being in a call (general stateless proxy forward). Is it possible to do 
this in asterisk using some tricks?

Regards,

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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[asterisk-users] AMI redirect from Queue to MeetMe

[Deka, Rajib IN MAA SL]

Hello List,

I have scenario as follows,


 1.  A call comes to queue.
 2.  Available agent will answer the call.
 3.  BridgeEvent wil be generated in AMI with channel1 and channel2.
 4.  Parse channel1 and channel two from the event and redirect them to a 
meetme room,

Dialplan,

Exten => 1234,1,MeetMe(1234,1dq)

But sometime it works and sometime one leg gets disconnected after redirection. 
Is it a bug to asterisk-1.6.2.13 ?

Regards,

Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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