Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from BlackBerry® on Airtel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Executing outbound dial number twice
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
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
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
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer and User Clarification
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 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording a meetme conference
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
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 -- next part -- An HTML attachment was scrubbed... URL: <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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 > -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
> > > 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from BlackBerry® on Airtel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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}-${UNIQUEID} and the default format is wav.
Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Recording a meetme conference
Here is a "sneaky" suggestion: start a local call that is recorded and have
it join the conference.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 12:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference
Danny,
You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded. But otherwise, it works fine.
I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.
Anybody with a recent 1.8 can tell me it works fine for them?
Mike
I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording a meetme conference
Danny,
You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded. But otherwise, it works fine.
I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it was already being recorded.
Anybody with a recent 1.8 can tell me it works fine for them?
Mike
I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?
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Re: [asterisk-users] Sangoma Card with 16E1 SS7 signaling
> Hi Team, > > i have been facing issues with sangoma card with 16 E1. > used LibSS7 > asterisk 1.6 > > with 8 E1 the links are stable , but moment i add another card of 8 E1 > for to support 16 E1. link keeps fluctuating > > any idea why ? > Your 16th channel may be mismatched with the network. Timeslot 16 is usually used for signaling. channels => 1-15,17 Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording a meetme conference
I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Recording a meetme conference
Hi,
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
Which is fine, but , but how do I change the default filename. Let's say I
want /tmp/recording.wav?
Mike
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Re: [asterisk-users] Recording a meetme conference
On Thu, Oct 27, 2011 at 11:53 AM, Mike wrote:
> I am trying to record a MeetMe conference, and this is what is relevant in
> the 1.8 manual:
>
>
>
> r - Record conference (records as MEETME_RECORDINGFILE using format
> MEETME_RECORDINGFORMAT. Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
>
>
>
> Which is fine, but , but how do I change the default filename. Let’s say I
> want /tmp/recording.wav?
There is not a rule against using Monitor() or MixMonitor() on the
entire channel. This might give you the control (format, filename) you
are looking for.
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[asterisk-users] Recording a meetme conference
Hi,
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${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
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI spans up without physical connections
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check which client access Asterisk using AMI
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tips & best practices for asterisk troubleshooting & parsing logs
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
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OPTIONS support for SDP
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
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/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent call monitoring
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
> >
> >
> > --
> >
> _
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com -- New to
> > Asterisk? Join us for a live introductory webinar
> every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> >
> _
> > -- Bandwidth and Colocation Provided by
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> Ishfaq Malik
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[asterisk-users] Unknown warning
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment
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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