Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Vinod Dharashive
Hi Patrick,

I did that, sangoma does not support Libss7 stack and they mentioned that 
Libss7 does work right with sangoma card with 16 e1.

Thanks
Vinod Dharashive
--Original Message--
From: Patrick Lists
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-user
ReplyTo: asterisk-user
Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
Sent: Oct 28, 2011 3:43 AM

On 10/27/2011 08:57 PM, Vinod Dharashive wrote:
> Hi Richard,
>
> Thanks for reply, how can I identify clocking issues on card. Manually calls 
> goes successfully on both the card. After 10 min signaling  links goes down.

Since you have a Sangoma card why don't you ask Sangoma support instead 
of asking Digium?

Patrick

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Re: [asterisk-users] Asterisk Executing outbound dial number twice

2011-10-27 Thread Patrick Lists

On 10/28/2011 01:02 AM, motty.cruz wrote:

Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice

Log

  == Using SIP RTP CoS mark 5
 -- Executing [912066604@sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
   == Extension Changed 4773[sipphones] new state InUse for Notify User 4701
 -- Executing [912066604@sipphones:2] Dial("SIP/4773-0003e920",
"SIP/att/xxx,80") in new stack


I still only see one Set and one Dial.

Regards,
Patrick

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Re: [asterisk-users] Asterisk Executing outbound dial number twice

2011-10-27 Thread isrlgb
Where do you see that ?
In the log you sent its setting the callerid and then dialing 

-Original Message-
From: "motty.cruz" 
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 27 Oct 2011 16:02:46 
To: 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] Asterisk Executing outbound dial number twice

Hello, 
I noticed Asterisk 1.8.4.1 execute number dial twice 

Log

 == Using SIP RTP CoS mark 5
-- Executing [912066604@sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
  == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 
-- Executing [912066604@sipphones:2] Dial("SIP/4773-0003e920",
"SIP/att/xxx,80") in new stack

Can you please help? 

Thanks, 
Motty


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[asterisk-users] Asterisk Executing outbound dial number twice

2011-10-27 Thread motty.cruz
Hello, 
I noticed Asterisk 1.8.4.1 execute number dial twice 

Log

 == Using SIP RTP CoS mark 5
-- Executing [912066604@sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
  == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 
-- Executing [912066604@sipphones:2] Dial("SIP/4773-0003e920",
"SIP/att/xxx,80") in new stack

Can you please help? 

Thanks, 
Motty


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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Patrick Lists

On 10/27/2011 08:57 PM, Vinod Dharashive wrote:

Hi Richard,

Thanks for reply, how can I identify clocking issues on card. Manually calls 
goes successfully on both the card. After 10 min signaling  links goes down.


Since you have a Sangoma card why don't you ask Sangoma support instead 
of asking Digium?


Patrick

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Re: [asterisk-users] Peer and User Clarification

2011-10-27 Thread Elliot Murdock
Hello,

Thank you for the response.

Does that mean when setting up a realtime system, one does not need to
specify both sippeers and sippusers?

Thanks,
Elliot

On Sun, Oct 23, 2011 at 12:18 PM, Alex Balashov
 wrote:
> All endpoints are peers, in the broad sense of "entities in sip.conf". This 
> includes phones, gateways, provider endpoints, etc.
>
> When a phone makes a call through an Asterisk server, it initiates a call leg 
> to Asterisk, which is matched to a sip.conf peer.  Asterisk then initiates a 
> second call leg through another sip.conf peer, and bridges the two legs 
> together.  Both are anchored by peers.
>
> The "type" of the peer (the type= setting) is a configuration detail that 
> changes some minor aspects of how the endpoint is treated, but whether the 
> type is "friend", "peer", etc. it's still a peer. They are largely the same.
>
> --
> This message was painstakingly thumbed out on my mobile, so apologies for 
> brevity, errors, and general sloppiness.
>
> 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 Oct 23, 2011, at 5:46 AM, Elliot Murdock  wrote:
>
>> Hello All,
>>
>> It seems from the Asterisk documentation, a User places phone calls
>> into the Asterisk server and a Peers accepts phone calls from the
>> Asterisk server.
>>
>> However, according to the document describing the "register =>"
>> command for sip.conf, it seems that Peers can in fact place calls into
>> an Asterisk system.  Is this correct and how is this working?
>>
>> Thanks,
>> Elliot
>>
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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread Mike
For future Google searches:

 

Ended up working with MixMonitor, but I had to remove the b option of
MixMonitor (record only when call is bridged).

 

I still haven`t figured out how to make the r option of MeetMe  function
properly, but I`ll give up since I have a working solution.

 

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Hi,

 

I thought of that, the problem is I have plenty of calls recorded, I want
the conf call ones to have the name ConfCall-whateverelse.wav

 

I'll get back to debugging and try to find why those aren't recorded but my
normal calls are. I am sure there is a reason, even if it escapes me at the
moment.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, October 27, 2011 1:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Here is a "sneaky" suggestion: start a local call that is recorded and have
it join the conference.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 12:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Danny,

 

You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded.  But otherwise, it works fine.

 

I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.

 

Anybody with a recent 1.8 can tell me it works fine for them?

 

Mike

 

 

 

 

I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?

 

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

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Re: [asterisk-users] Check which client access Asterisk using AMI

2011-10-27 Thread Danny Nicholas
If you only want connects and disconnects, verbose should suffice

[Oct 27 14:39:55] VERBOSE[3192] logger.c:   == Parsing
'/etc/asterisk/manager.conf': [Oct 27 14:39:55] VERBOSE[3192] logger.c:
Found

[Oct 27 14:39:55] VERBOSE[3192] logger.c:   == Manager 'debsman' logged on
from 192.168.23.172

[Oct 27 14:39:56] WARNING[3192] chan_sip.c: No such host: 104

[Oct 27 14:39:56] NOTICE[3192] channel.c: Unable to request channel SIP/104

[Oct 27 14:39:58] VERBOSE[3192] logger.c:   == Manager 'debsman' logged off
from 192.168.23.172

[Oct 27 14:40:22] VERBOSE[13080] logger.c: -- Remote UNIX connection

[Oct 27 14:48:30] VERBOSE[3238] logger.c: -- Remote UNIX connection
disconnected

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Thursday, October 27, 2011 2:07 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Check which client access Asterisk using AMI

 


Is there other way around doing it instead of enabling debug and verbose
from logger.conf?






Message: 1
Date: Thu, 27 Oct 2011 10:36:08 -0400
From: Ahmed Munir 
Subject: [asterisk-users] Check which client access Asterisk using AMI
To: asterisk-users@lists.digium.com
Message-ID:
   
Content-Type: text/plain; charset="iso-8859-1"

Hi,

In manager.conf file I created a user profile by which clients can access
Asterisk server as listed below;


[cbusapp]
secret = cbus123
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate



Using above configuration clients are successfully access the asterisk and
forward its parameters to asterisk. The thing I would like to know how can I
keep track from which client does asterisk receives request from? Like
client A, B and C need to know from which clients the request was made to
asterisk.

--
Regards,

Ahmed Munir Chohan
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<http://lists.digium.com/pipermail/asterisk-users/attachments/20111027/ff89e
a1a/attachment.html>

--

Message: 2
Date: Thu, 27 Oct 2011 09:39:43 -0500
From: "Danny Nicholas" 
Subject: Re: [asterisk-users] Check which client access Asterisk using
   AMI
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
   
Message-ID: <006e01cc94b6$443a6c60$ccaf4520$@debsinc.com>
Content-Type: text/plain; charset="us-ascii"

This information might be in /var/log/asterisk/messages or /v/l/a/full.  If
not, you can change the logging and get it there (turn on debug in one of
them) (/etc/asterisk/logger.conf)




-- 
Regards,

Ahmed Munir Chohan



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Re: [asterisk-users] Check which client access Asterisk using AMI

2011-10-27 Thread Ahmed Munir
Is there other way around doing it instead of enabling debug and verbose
from logger.conf?





> Message: 1
> Date: Thu, 27 Oct 2011 10:36:08 -0400
> From: Ahmed Munir 
> Subject: [asterisk-users] Check which client access Asterisk using AMI
> To: asterisk-users@lists.digium.com
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> In manager.conf file I created a user profile by which clients can access
> Asterisk server as listed below;
>
>
> [cbusapp]
> secret = cbus123
> deny=0.0.0.0/0.0.0.0
> permit=192.168.1.0/255.255.255.0
> read = system,call,log,verbose,command,agent,user,originate
> write = system,call,log,verbose,command,agent,user,originate
>
>
>
> Using above configuration clients are successfully access the asterisk and
> forward its parameters to asterisk. The thing I would like to know how can
> I
> keep track from which client does asterisk receives request from? Like
> client A, B and C need to know from which clients the request was made to
> asterisk.
>
> --
> Regards,
>
> Ahmed Munir Chohan
> -- next part ------
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20111027/ff89ea1a/attachment.html
> >
>
> --
>
> Message: 2
> Date: Thu, 27 Oct 2011 09:39:43 -0500
> From: "Danny Nicholas" 
> Subject: Re: [asterisk-users] Check which client access Asterisk using
>AMI
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> Message-ID: <006e01cc94b6$443a6c60$ccaf4520$@debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> This information might be in /var/log/asterisk/messages or /v/l/a/full.  If
> not, you can change the logging and get it there (turn on debug in one of
> them) (/etc/asterisk/logger.conf)
>
>
>
> --
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Vinod Dharashive
Hi Richard,

Thanks for reply, how can I identify clocking issues on card. Manually calls 
goes successfully on both the card. After 10 min signaling  links goes down.

Thanks
Vinod dharashive
--Original Message--
From: Richard Mudgett
To: vdharash...@gmail.com
To: asterisk-user
Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
Sent: Oct 27, 2011 11:34 PM

> > > Hi Team,
> > >
> > > i have been facing issues with sangoma card with 16 E1.
> > > used LibSS7
> > > asterisk 1.6
> > >
> > > with 8 E1 the links are stable , but moment i add another card of 8
> > > E1
> > > for to support 16 E1. link keeps fluctuating
> > >
> > > any idea why ?
> > >
> > Your 16th channel may be mismatched with the network. Timeslot 16
> > is usually used for signaling. channels => 1-15,17
> 
> The link is up on 16 th channel. My objective is to have 16 E1 to be
> configure on single machine with two 8 port sangoma card. Which is
> problem I am facing. Please let me know if you have any solution.
> 
Sounds like it could be a clocking issue between the two cards then.
Everything needs to use the same clock source.

Richard


Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Vinod Dharashive
Hi stefan,

Do you have any configuration file for 16 E1 of system.conf and wanpipeX.conf. 
Can u please share with me.

Thanks
Vinod dharashive.


--Original Message--
From: Stefan Schmidt
To: vdharash...@gmail.com
To: asterisk-user
Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
Sent: Oct 27, 2011 11:23 PM

Am 27.10.2011 19:29, schrieb Vinod Dharashive:
> Hi Richard,
> 
> The link is up on 16 th channel. My objective is to have 16 E1 to be 
> configure on single machine with two 8 port sangoma card. Which is problem I 
> am facing. Please let me know if you have any solution.
> 
> Thanks
> Vinod dharashive
> 
> 

Hi vinod,

i have never tried to put two A108 cards into one server but it sounds
like an IRQ Problem to me. you should check all system relevant
information (lsmod, lspci,dmesg) if you can see any kernel errors with this.

it could also be a span identification problem that chan_dahdi recognize
the second card also beginning with span 1 and not 9 or something like this.


btw i dont think that you will be happy with asterisk 1.6 and handling
496 concurrent calls over ss7

best regards

Stefan



Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Richard Mudgett
> > > Hi Team,
> > >
> > > i have been facing issues with sangoma card with 16 E1.
> > > used LibSS7
> > > asterisk 1.6
> > >
> > > with 8 E1 the links are stable , but moment i add another card of 8
> > > E1
> > > for to support 16 E1. link keeps fluctuating
> > >
> > > any idea why ?
> > >
> > Your 16th channel may be mismatched with the network. Timeslot 16
> > is usually used for signaling. channels => 1-15,17
> 
> The link is up on 16 th channel. My objective is to have 16 E1 to be
> configure on single machine with two 8 port sangoma card. Which is
> problem I am facing. Please let me know if you have any solution.
> 
Sounds like it could be a clocking issue between the two cards then.
Everything needs to use the same clock source.

Richard

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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Stefan Schmidt
Am 27.10.2011 19:29, schrieb Vinod Dharashive:
> Hi Richard,
> 
> The link is up on 16 th channel. My objective is to have 16 E1 to be 
> configure on single machine with two 8 port sangoma card. Which is problem I 
> am facing. Please let me know if you have any solution.
> 
> Thanks
> Vinod dharashive
> 
> 

Hi vinod,

i have never tried to put two A108 cards into one server but it sounds
like an IRQ Problem to me. you should check all system relevant
information (lsmod, lspci,dmesg) if you can see any kernel errors with this.

it could also be a span identification problem that chan_dahdi recognize
the second card also beginning with span 1 and not 9 or something like this.


btw i dont think that you will be happy with asterisk 1.6 and handling
496 concurrent calls over ss7

best regards

Stefan


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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Vinod Dharashive
Hi Richard,

The link is up on 16 th channel. My objective is to have 16 E1 to be configure 
on single machine with two 8 port sangoma card. Which is problem I am facing. 
Please let me know if you have any solution.

Thanks
Vinod dharashive


--Original Message--
From: Richard Mudgett
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-user
ReplyTo: asterisk-user
Subject: Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
Sent: Oct 27, 2011 9:33 PM

> Hi Team,
> 
> i have been facing issues with sangoma card with 16 E1.
> used LibSS7
> asterisk 1.6
> 
> with 8 E1 the links are stable , but moment i add another card of 8 E1
> for to support 16 E1. link keeps fluctuating
> 
> any idea why ?
> 
Your 16th channel may be mismatched with the network.  Timeslot 16
is usually used for signaling.  channels => 1-15,17

Richard

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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread Mike
Hi,

 

I thought of that, the problem is I have plenty of calls recorded, I want
the conf call ones to have the name ConfCall-whateverelse.wav

 

I'll get back to debugging and try to find why those aren't recorded but my
normal calls are. I am sure there is a reason, even if it escapes me at the
moment.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, October 27, 2011 1:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Here is a "sneaky" suggestion: start a local call that is recorded and have
it join the conference.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 12:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Danny,

 

You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded.  But otherwise, it works fine.

 

I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.

 

Anybody with a recent 1.8 can tell me it works fine for them?

 

Mike

 

 

 

 

I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?

 

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread Danny Nicholas
Here is a "sneaky" suggestion: start a local call that is recorded and have
it join the conference.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 12:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference

 

Danny,

 

You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded.  But otherwise, it works fine.

 

I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.

 

Anybody with a recent 1.8 can tell me it works fine for them?

 

Mike

 

 

 

 

I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?

 

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread Mike
Danny,

 

You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded.  But otherwise, it works fine.

 

I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.

 

Anybody with a recent 1.8 can tell me it works fine for them?

 

Mike

 

 

 

 

I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?

 

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Richard Mudgett
> Hi Team,
> 
> i have been facing issues with sangoma card with 16 E1.
> used LibSS7
> asterisk 1.6
> 
> with 8 E1 the links are stable , but moment i add another card of 8 E1
> for to support 16 E1. link keeps fluctuating
> 
> any idea why ?
> 
Your 16th channel may be mismatched with the network.  Timeslot 16
is usually used for signaling.  channels => 1-15,17

Richard

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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread Danny Nicholas
I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Recording a meetme conference

 

Hi,

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

 

 

 

Mike

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Re: [asterisk-users] Recording a meetme conference

2011-10-27 Thread David Backeberg
On Thu, Oct 27, 2011 at 11:53 AM, Mike  wrote:
> I am trying to record a MeetMe conference, and this is what is relevant in
> the 1.8 manual:
>
>
>
> r - Record conference (records as MEETME_RECORDINGFILE using format
> MEETME_RECORDINGFORMAT. Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
>
>
>
> Which is fine, but , but how do I change the default filename. Let’s say I
> want /tmp/recording.wav?

There is not a rule against using Monitor() or MixMonitor() on the
entire channel. This might give you the control (format, filename) you
are looking for.

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[asterisk-users] Recording a meetme conference

2011-10-27 Thread Mike
Hi,

 

I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:

 

r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.

 

Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?

 

 

 

 

Mike

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[asterisk-users] Sangoma Card with 16E1 SS7 signaling

2011-10-27 Thread Vinod Dharashive
Hi Team,

i have been facing issues with sangoma card with 16 E1.
used LibSS7
asterisk 1.6

with 8 E1 the links are stable , but moment i add another card of 8 E1 for
to support 16 E1. link keeps fluctuating

any idea why ?

Please help

Thanks
Vinod Dharashive
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[asterisk-users] DAHDI spans up without physical connections

2011-10-27 Thread Ron Bergin
We are in the process of rebuilding our servers and upgrading asterisk and
zaptel/dahdi.

The server I'm currently working on was:
OS: Fedora fc9
libpri 1.4.10.2
Zaptel 1.4.12.1
Asterisk: 1.4.27

After rebuild:
OS: CentOS 5.7 (Final) installed via Spacewalk
libpri: 1.4.12
DAHDI: dahdi-linux-complete-2.5.0.2+2.5.0.2.
Asterisk 1.8.6

Digium card TE420:
pci::09:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen)

dahdi spans are configured as:
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs

fxoks=1-24
fxoks=25-48
fxoks=49-72

No errors during build process, but the dahdi spans are showing up even
though there are no cables connected to the card's T1 ports.

We have rebuilt libpri and dahdi multiple times and with various
combinations of versions (libpri 1.4.11.3 and 14..12 and dahdi 2.3 and
2.5), but always have the same results.

One possible clue would be this error in dmesg (which I don't fully
understand):
WARNING: FALC framer not intialized in compatibility mode.
ERROR: FALC framer version is unknown (VSTR = 0x00).
FALC version: , Board ID: 00

Has anyone come across this problem and/or can point me in the right
direction to fix it?

Thanks
-- 
Ron Bergin
Fry's Electronics, Inc





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Re: [asterisk-users] Check which client access Asterisk using AMI

2011-10-27 Thread Danny Nicholas
This information might be in /var/log/asterisk/messages or /v/l/a/full.  If
not, you can change the logging and get it there (turn on debug in one of
them) (/etc/asterisk/logger.conf)

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Thursday, October 27, 2011 9:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Check which client access Asterisk using AMI

 

Hi,

In manager.conf file I created a user profile by which clients can access
Asterisk server as listed below;


[cbusapp]
secret = cbus123
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate



Using above configuration clients are successfully access the asterisk and
forward its parameters to asterisk. The thing I would like to know how can I
keep track from which client does asterisk receives request from? Like
client A, B and C need to know from which clients the request was made to
asterisk.

-- 
Regards,

Ahmed Munir Chohan



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[asterisk-users] Check which client access Asterisk using AMI

2011-10-27 Thread Ahmed Munir
Hi,

In manager.conf file I created a user profile by which clients can access
Asterisk server as listed below;


[cbusapp]
secret = cbus123
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate



Using above configuration clients are successfully access the asterisk and
forward its parameters to asterisk. The thing I would like to know how can I
keep track from which client does asterisk receives request from? Like
client A, B and C need to know from which clients the request was made to
asterisk.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Tips & best practices for asterisk troubleshooting & parsing logs

2011-10-27 Thread Anton Kvashenkin
Capture pcap with tshark or tcpdump for the future analysis with wireshark.
Ngrep is also handy tool for captaring, say, INVITE. You can use grep like
this: tail -f /var/log/asterisk/full | egrep --color -w
'chan_sip.*SIP/911|pbx.*SIP/911'
Interesting technique from Astresk Cookbook, "Debugging dialplan with
Verbose()
http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html

2011/10/27 Sammy Govind 

> It was a challenge to read through all the interesting experience you've
> shared over here. I don't know what others may be using for parsing the logs
> beautifully and make them usable. What I would recommend you at the very
> beginning ,since you mentioned using egrep, is figure out the Channel
> identifier string from the logs for a particular call. That's underlined
> below for you.
>
> [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3]
>> System("*Local/s@tc-maint-2496,2*","/var/lib/asterisk/bin/schedtc.php 60
>> /var/spool/asterisk/outgoing 0") in new stack
>
>
> Once you Figure out this part use egrep tool and you'll end up seeing only
> the data related to this particular call.
>
> More advanced tool or techniques may involve setting up a central logging
> server where all the other servers deposit their logs and use monitoring
> tools like swatch, splunk, zabbix etc etc etc to parse the logs for you and
> generate alerts.
>
> I haven't came across any Asterisk-specific log parser utility so far.
> Honestly, I never needed one.
>
> On Thu, Oct 27, 2011 at 5:16 AM, Douglas Mortensen <
> d...@impalanetworks.com> wrote:
>
>> Hello all,
>>
>> I have been running asterisk systems since summer of 2008. I do not claim
>> to be an expert. But I have worked through many issues during this period. I
>> have setup & manage 5 systems, which serve 6 companies total (and of course
>> process calls for all of the people they do business with).
>>
>> I have always been happy with asterisk (well, obviously less happy during
>> the problem times... :-). And I continue to prefer to us it. However, if I
>> could name the one largest struggle that I have with asterisk, it is the
>> facilities that it provides for troubleshooting issues & parsing logs.
>>
>> I am hoping that someone on this mailing list can help me to realize how
>> ignorant I really am, and how much time I have wasted parsing, "grep"ping &
>> "less"ing logs manually. I am hoping that one of you can help me "see the
>> light". If so, I would be most grateful.
>>
>> Specifically, here are the challenges I encounter, which I would
>> desperately appreciate help with:
>>
>> Here's an example scenario:
>>
>> A customer calls me & says that a call just came in & some of their
>> wireless DECT phones (I know, trouble already :-) didn't ring, while
>> others did. I tell the customer that I'll start looking into the problem
>> immediately.
>>
>> I am using AsteriskNOW with asterisk 1.6. So I SSH into the system & cd to
>> /var/log/asterisk & start looking at the "full" log via "less". We have
>> configured the bulk of our system via FreePBX 2.9. Inbound calls are routed
>> first to a time condition which checks whether it is after hours. If it is
>> not afterhours, then are then routed to a queue, which rings all phones (4
>> wireless DECT phones on 1 DECT wireless server that registers the SIP
>> extensions on behalf of its 4 phones, and 4 more wireless DECT phones on
>> their own wireless server configured the same, and an ATA connected to a
>> paging amp that rings a loud speaker). From there, someone typically will
>> answer the call. Often times they then transfer the call to another
>> extension. However, sometimes no one answers the call, and it winds up going
>> to VM.
>>
>> From the logging aspect of asterisk, it has usually felt like I am
>> trudging through a swampy marsh trying to put the bits & pieces together.
>> The challenge I've seen is that the above scenario can actually consist of
>> multiple SIP calls w/ different legs. I *think* (but am not 100% sure) that
>> often times a call can be handed off from 1 asterisk process to another. The
>> result is that "grep"ping by the asterisk process ID shown after the VERBOSE
>> (or NOTICE or DEBUG section [see below]), I don't actually get to see the
>> full sequence of events in following  all logging that is relevant to that
>> phone call.
>>
>> [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3]
>> System("Local/s@tc-maint-2496,2","/var/lib/asterisk/bin/schedtc.php 60
>> /var/spool/asterisk/outgoing 0") in new stack
>>
>> Then on a busy asterisk system, if I filter by the process id, the one
>> process that starts handling the call originally, may wind up immediately
>> taking on another totally unrelated call after handing the initial call off
>> to another process. If I am not extremely careful, I may wind up mistaking
>> the log lines for the 2nd call, as being a part of the 1st call, and then
>> I'm totally barking up the wrong tre

Re: [asterisk-users] Unknown warning

2011-10-27 Thread Ishfaq Malik
For the record I think I got to the bottom of this.

I'm still using 1.8.3 and this issue is occurring where SIP channels get
stuck

https://issues.asterisk.org/jira/browse/ASTERISK-18225

I think the warnings are these stuck channels trying to reconnect. 

On Thu, 2011-10-27 at 11:15 +0100, Ishfaq Malik wrote:
> Is it anything to worry about? there are about 8 a second happening.
> 
> On Thu, 2011-10-27 at 06:10 -0400, Alex Balashov wrote:
> > It means Asterisk is enqueueing a failed reinvite for retransmission.
> > 
> > On 10/27/2011 06:04 AM, Ishfaq Malik wrote:
> > 
> > > Hi
> > >
> > > Can anyone shed some light on what this warning means?
> > >
> > > chan_sip.c:19184 handle_response_invite: just did sched_add
> > > waitid(1223301) for sip_reinvite_retry for dialog
> > > 3c46ab7f1762-8nxnhonpfcgr in handle_response_invite
> > >
> > > I've had a good look online but can't find a decent answer.
> > >
> > > Thanks in advance
> > >
> > > Ish
> > 
> > 
> 

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] OPTIONS support for SDP

2011-10-27 Thread J.R. Pauley
I have been sending OPTIONS requests 1) programatically (my own code),
 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes
in sip.conf. The trouble is I do not see anything except an ACK 200 come
back from endpoints and it does not contain any SDP/codec info. . My goal is
to determine audio and video codec capability in advance of a call INVITE. I
notice in both 2 and 3 examples the Asterisk generated OPTIONS does not
specify any ACCEPT header (ie Accept=application/sdp). I was thinking maybe
that is why I don't get any SDP coming back. The rfc says the ACCEPT SHOULD
be present so I'm thinking that is a Asterisk bug perhaps. In example 1 My
own UAC code generated OPTIONS includes the Accept header yet still I see no
SDP coming back from endpoints. I have tried using X-lite and PhonerLite
softclients. I'm hoping there is a simple explanation or something I can
do.

Is Anyone able to query codec capability for any endpoints outside of a
normal INVITE? I would like to know how you do so.

Below is excerpt from the automatic OPTIONS query I see in the sip logs from
setting verify=true. No Accept header. Does anyone believe that to be the
problem? Notice the response has content length=0 and no SDP. Any ideas
appreciated

OPTIONS sip:991@192.168.1.4:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169

Max-Forwards: 70

From: "asterisk" ;tag=as1fd2a50c

To: 

Contact: 

Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.6.0

Date: Mon, 24 Oct 2011 19:14:47 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH

Supported: replaces, timer

Content-Length: 0


<--- SIP read from UDP:192.168.1.4:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169

From: "asterisk" ;tag=as1fd2a50c

To: ;tag=003d3418e2fce011b081701a0413e3f3

Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060

CSeq: 102 OPTIONS

Contact: 

Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE

Server: SIPPER for PhonerLite

Content-Length: 0
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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Ishfaq Malik
On Thu, 2011-10-27 at 06:32 -0400, Alex Balashov wrote:
> On 10/27/2011 06:30 AM, Ishfaq Malik wrote:
> > On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote:
> >> On 10/27/2011 06:15 AM, Ishfaq Malik wrote:
> >>
> >>> Is it anything to worry about? there are about 8 a second
> >>> happening.
> >>
> >> Maybe.  Can't say without more data/context/packet capture/etc.
> >>
> > Well it doesn't seem to be having too big an impact on load or cpu
> > usage.
> >
> > Any pointers on where I can do any further digging?
> 
> Take a capture and see what kind of SIP flow is causing it:
> 
> tcpdump -i any -A -w capture.pcap -s 0 -n 'udp port 5060'
> 
> After you're reasonably sure a few of those errors have occurred, hit 
> Ctrl+C to stop the capture.  Then, open up capture.pcap in Wireshark 
> and see what's what, and/or get someone who knows a lot about SIP to 
> help you.
> 
That's plenty for me to go on, thanks a lot for your help.
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Alex Balashov

On 10/27/2011 06:30 AM, Ishfaq Malik wrote:

On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote:

On 10/27/2011 06:15 AM, Ishfaq Malik wrote:


Is it anything to worry about? there are about 8 a second
happening.


Maybe.  Can't say without more data/context/packet capture/etc.


Well it doesn't seem to be having too big an impact on load or cpu
usage.

Any pointers on where I can do any further digging?


Take a capture and see what kind of SIP flow is causing it:

   tcpdump -i any -A -w capture.pcap -s 0 -n 'udp port 5060'

After you're reasonably sure a few of those errors have occurred, hit 
Ctrl+C to stop the capture.  Then, open up capture.pcap in Wireshark 
and see what's what, and/or get someone who knows a lot about SIP to 
help you.


--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Ishfaq Malik
On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote:
> On 10/27/2011 06:15 AM, Ishfaq Malik wrote:
> 
> > Is it anything to worry about? there are about 8 a second
> > happening.
> 
> Maybe.  Can't say without more data/context/packet capture/etc.
> 
Well it doesn't seem to be having too big an impact on load or cpu
usage. 

Any pointers on where I can do any further digging?

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PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Alex Balashov

On 10/27/2011 06:15 AM, Ishfaq Malik wrote:


Is it anything to worry about? there are about 8 a second
happening.


Maybe.  Can't say without more data/context/packet capture/etc.

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Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
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Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Ishfaq Malik
Is it anything to worry about? there are about 8 a second happening.

On Thu, 2011-10-27 at 06:10 -0400, Alex Balashov wrote:
> It means Asterisk is enqueueing a failed reinvite for retransmission.
> 
> On 10/27/2011 06:04 AM, Ishfaq Malik wrote:
> 
> > Hi
> >
> > Can anyone shed some light on what this warning means?
> >
> > chan_sip.c:19184 handle_response_invite: just did sched_add
> > waitid(1223301) for sip_reinvite_retry for dialog
> > 3c46ab7f1762-8nxnhonpfcgr in handle_response_invite
> >
> > I've had a good look online but can't find a decent answer.
> >
> > Thanks in advance
> >
> > Ish
> 
> 

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Unknown warning

2011-10-27 Thread Alex Balashov

It means Asterisk is enqueueing a failed reinvite for retransmission.

On 10/27/2011 06:04 AM, Ishfaq Malik wrote:


Hi

Can anyone shed some light on what this warning means?

chan_sip.c:19184 handle_response_invite: just did sched_add
waitid(1223301) for sip_reinvite_retry for dialog
3c46ab7f1762-8nxnhonpfcgr in handle_response_invite

I've had a good look online but can't find a decent answer.

Thanks in advance

Ish



--
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] Concurrent call monitoring

2011-10-27 Thread Ishfaq Malik
Thanks for the tips, I'll try this method now I know.

On Tue, 2011-10-25 at 18:52 +0500, Sammy Govind wrote:
> We used Mix of Nagios, Zabbix, OpenNMS. Best one for this was Zabbix.
> 
> On Tue, Oct 25, 2011 at 6:49 PM, Ishfaq Malik 
> wrote:
> Which monitoring tool were you using?
> 
> 
> On Tue, 2011-10-25 at 18:46 +0500, Sammy Govind wrote:
> > I wrote my own shell scripts to collect "core show calls"
> value from
> > asterisk and then push the filtered value to an opensource
> monitoring
> > tool. That worked perfectly well.
> >
> >
> > #!/usr/bin/perl -w
> > use strict;
> > open(LINE, 'asterisk -rx "core show channels"|');
> > my ($chans, $calls, $line)=(0,0,undef);
> > while ($line = )
> > {
> > $calls = $1 if ($line =~ /^(\d+) active call/);
> > }
> > close(LINE);
> > printf $calls;
> >
> >
> >
> > On Tue, Oct 25, 2011 at 6:40 PM, Danny Nicholas
> 
> > wrote:
> > The "Simplest" method of seeing the number of
> concurrent calls
> > is "service
> > asterisk status".  If I understand question two,
>  asterisk -rx
> > " core show
> > channels verbose" is probably your best bet.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On
> Behalf Of
> > Ishfaq Malik
> > Sent: Tuesday, October 25, 2011 8:29 AM
> > To: Asterisk Users
> > Subject: [asterisk-users] Concurrent call monitoring
> >
> >
> > Hi
> >
> > What are people using to monitor the concurrent
> number of
> > calls at any given
> > time?
> >
> > Also, is there any good way of monitoring concurrent
> inbound
> > and outbound
> > calls so that we can see the 2 different numbers?
> >
> > Thanks in advance
> >
> > Ish
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office:   0161 660 3062
> >
> >
> > --
> >
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> >
> >
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> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
> 
> Office:   0161 660 3062
> 
> 
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> 

[asterisk-users] Unknown warning

2011-10-27 Thread Ishfaq Malik
Hi

Can anyone shed some light on what this warning means?

chan_sip.c:19184 handle_response_invite: just did sched_add
waitid(1223301) for sip_reinvite_retry for dialog
3c46ab7f1762-8nxnhonpfcgr in handle_response_invite

I've had a good look online but can't find a decent answer.

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment

2011-10-27 Thread Samuel Sappa
Dear all,
I'm still having trouble using asterisk with the grandstream gxw4108,
in the gxw4108 I'm using 1 stage dialing in the profile1 I already
type my asterisk server address 192.168.14.80 and my grandstream IP is
192.168.101.184

here's my asterisk config files

SIP.CONF
[1401]
type = friend
username = 1401
secret = 1401
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes

[1402]
type = friend
username = 1402
secret = 1402
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes

EXTENTION.CONF
[kantor-mtx]
exten => 1401,1,Dial(SIP/1401,60)
exten => 1402,1,Dial(SIP/1402,60)
exten => _NXXXNXXX,1,Dial(SIP/${EXTEN}@1401)
exten => _0813,1,Dial(SIP/${EXTEN}@1402)
exten => _0812,1,Dial(SIP/${EXTEN}@1402)

with this configuration files the result :
1.Sometime when i'm dialing to the PSTN line, not the PSTN ringing,
instead the extention is ringing
2.when I restart both asterisk and gxw4108, it's succes when dialing
to PSTN, but when I try to dial the extention it's seems dialing to
PSTN

(I also configuring to separate my incoming and outgoing call but
still the same error occurs)

my question is :
1.is it my gateway is broken or malfunction ?
2.When testing this configuration I'm using only 2 PSTN line instead
of 8 lines provide by the gateway, is this can cause problem in my
configuration ?
3.Or there is some other configuration for my asterisk and gxw4108,
instead the one I'm using would kindly to share in here

Thank you very much for your guidance and sorry if my English is bad
-- 
Regards
Samuel Sappa,

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