Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 1:49 AM, Faisal Hanif  wrote:

> You have to provide channel ID to command like “channel request hangup
> SIP/12316156-sad4d46a5”.
>
> **
>

Thanks, but "all" is also a valid keyword according to the documentation. I
think there are some bugs associated with hung channels. Nothing seems to
work when a channel is hung in that state. hanging up is not working, nor
the AMI is working in providing status etc. and when I'm on the CLI, even
"core stop now" doesn't work and it hands the CLI.

Something is majorly wrong. I'm going to upgrade the version to 1.8.4.4 and
see what happens


>  **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
> *Sent:* Wednesday, July 06, 2011 9:50 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+
> CPU with No calls on the system
>
> ** **
>
> ** **
>
> On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk 
> wrote:
>
> On the CLI write: sip show channels
>
> If there are lots of bye channels you have the same problem than me.
> I've tried waiting with the call generator -sipp- and channels
> finished when there are a few. But they're not ending faster enough
> when I send lots of concurrent calls.
>
> Elder
>
> Hi,
>
> thanks for the response. yeah I'd checked that before and I only have 2
> dialogs which seem to be part of the same call that are just sitting there
> and I can't seem to get them to hang up by typing "channel request hangup
> all" . I even tried sending a Hangup by connecting on the AMI but that
> doesn't seem to be doing anything either. So this channel is sitting there
> in the 'BYE' state. 
>
> Is there anyway of clearing them without having to reload/restart Asterisk?
> I want to see if that's the cause of the CPU usage and I'll lose that if I
> restart Asterisk.
>
> Thanks
>
> ** **
>
>  
>
> 2011/7/5, A E [Gmail] :
>
> > hello people,
> >
> > I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
> some
> > reason I have noticed that only after a few test calls, the asterisk
> process
> > is running between 95% - 99.9% CPU when there's absolutely nothing on the
> > system. This is a clean Asterisk system in an internal network with
> nothing
> > else on it with no calls on it but it's still sitting with 96% CPU.
> >
> > I'm not a developer so not that ept with using debug tools etc to figure
> out
> > why it's doing that. Could anyone please tell me how I can figure out why
> > it's doing this and/or help debug this. Makes no sense for it to be using
> > CPU with nothing happening on the system
> >
> > Thanks
> >
>
> --
> Enviado desde mi dispositivo móvil
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
> ** **
>
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[Faisal Hanif]

You have to provide channel ID to command like “channel request hangup
SIP/12316156-sad4d46a5”.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Wednesday, July 06, 2011 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU
with No calls on the system

 

 

On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk 
wrote:

On the CLI write: sip show channels

If there are lots of bye channels you have the same problem than me.
I've tried waiting with the call generator -sipp- and channels
finished when there are a few. But they're not ending faster enough
when I send lots of concurrent calls.

Elder

Hi,

thanks for the response. yeah I'd checked that before and I only have 2
dialogs which seem to be part of the same call that are just sitting there
and I can't seem to get them to hang up by typing "channel request hangup
all" . I even tried sending a Hangup by connecting on the AMI but that
doesn't seem to be doing anything either. So this channel is sitting there
in the 'BYE' state. 

Is there anyway of clearing them without having to reload/restart Asterisk?
I want to see if that's the cause of the CPU usage and I'll lose that if I
restart Asterisk.

Thanks

 

 

2011/7/5, A E [Gmail] :

> hello people,
>
> I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
> reason I have noticed that only after a few test calls, the asterisk
process
> is running between 95% - 99.9% CPU when there's absolutely nothing on the
> system. This is a clean Asterisk system in an internal network with
nothing
> else on it with no calls on it but it's still sitting with 96% CPU.
>
> I'm not a developer so not that ept with using debug tools etc to figure
out
> why it's doing that. Could anyone please tell me how I can figure out why
> it's doing this and/or help debug this. Makes no sense for it to be using
> CPU with nothing happening on the system
>
> Thanks
>

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Re: [asterisk-users] Blind Transfer Connected

[Olivier]

2011/7/6 Nikhil 

> **
> Hi
> Below is the comment that written in chan_sip.c(handle_request_refer)
> file of asterisk .In RFC also mentioned that if blind transfer failed call
> should connect back, some of phones support this(If received refer) like
> cisco,polycom and etc.
>
>  \par Blind transfers
> The transferor provides the transferee
> with the transfer targets contact. The signalling between
> transferer or transferee should not be cancelled, so the
> call is recoverable if the transfer target can not be reached
> by the transferee.
>

My understanding of this is :
"If transfer target (ie phone C) rings, then transfer target HAS BEEN
reached so the above statement do not apply".



>
> In this case, Asterisk receives a TRANSFER from
> the transferor, thus is the transferee. We should
> try to set up a call to the contact provided
> and if that fails, re-connect the current session.
> If the new call is set up, we issue a hangup.
> In this scenario, we are following section 5.2
> in the SIP CC Transfer draft. (Transfer without
> a GRUU)
>
>
>
> In asterisk comment is written correct but it is not working.
>
> Thanks
> Nikhil
>
> On 07/05/2011 09:44 PM, Kevin P. Fleming wrote:
>
> On 07/05/2011 01:54 AM, Olivier wrote:
>
>
>
> 2011/7/5 Nikhil   >
>
> Hi all
> In asterisk if blind transfer failed ,call is not connecting back .
>
>
> For Eg:
> A make call to B through asterisk,then B transfer the call to C.
> If C did not answer the call ,A  and B Call should connect back.
>
> IMHO, blind tranfer definition is to NOT connect A and B back
>
>
> That is correct, and is why it's called a 'blind' transfer; the
> transferring party does not know or care what happens to the call after
> effecting the transfer.
>
>
>
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> _
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Re: [asterisk-users] Couldn't call Agent and segfault

[Faisal Hanif]

If the problem always related to some specific module then try clean
recompiling asterisk if it is with random modules then check you system RAM.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina
Berretta
Sent: Wednesday, July 06, 2011 1:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Couldn't call Agent and segfault

 

Hi folks!

I´m having the following problem:

I get the following messages, asterisk get automatically reloaded and agents
log out once or twice a day, randomly.

[Jul 4 11:36:25] VERBOSE[30004] app_queue.c: -- Couldn't call Agent/2002
[Jul 4 11:36:29] VERBOSE[30320] logger.c: Asterisk Event Logger Started
/var/log/asterisk/event_log
[Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing
'/etc/asterisk/asterisk.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: ==
Found
[Jul 4 11:36:29] VERBOSE[30320] loader.c: Asterisk Dynamic Loader Starting:
[Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing
'/etc/asterisk/modules.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: ==
Found
[Jul 4 11:36:29] NOTICE[30320] loader.c: 2 modules will be loaded.

Also I get a segfault in /var/log/messages.
Any help will be appreciated! 

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Re: [asterisk-users] Blind Transfer Connected

[Nikhil]

Hi
Below is the comment that written in 
chan_sip.c(handle_request_refer) file of asterisk .In RFC also mentioned 
that if blind transfer failed call should connect back, some of phones 
support this(If received refer) like cisco,polycom and etc.


\par Blind transfers
The transferor provides the transferee
with the transfer targets contact. The signalling between
transferer or transferee should not be cancelled, so the
call is recoverable if the transfer target can not be reached
by the transferee.

In this case, Asterisk receives a TRANSFER from
the transferor, thus is the transferee. We should
try to set up a call to the contact provided
and if that fails, re-connect the current session.
If the new call is set up, we issue a hangup.
In this scenario, we are following section 5.2
in the SIP CC Transfer draft. (Transfer without
a GRUU)



In asterisk comment is written correct but it is not working.

Thanks
Nikhil

On 07/05/2011 09:44 PM, Kevin P. Fleming wrote:

On 07/05/2011 01:54 AM, Olivier wrote:



2011/7/5 Nikhil mailto:d.nik...@cem-solutions.net>>

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


For Eg:
A make call to B through asterisk,then B transfer the call to C.
If C did not answer the call ,A  and B Call should connect back.

IMHO, blind tranfer definition is to NOT connect A and B back


That is correct, and is why it's called a 'blind' transfer; the 
transferring party does not know or care what happens to the call 
after effecting the transfer.




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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk wrote:

> On the CLI write: sip show channels
>
> If there are lots of bye channels you have the same problem than me.
> I've tried waiting with the call generator -sipp- and channels
> finished when there are a few. But they're not ending faster enough
> when I send lots of concurrent calls.
>
> Elder
>
> Hi,
thanks for the response. yeah I'd checked that before and I only have 2
dialogs which seem to be part of the same call that are just sitting there
and I can't seem to get them to hang up by typing "channel request hangup
all" . I even tried sending a Hangup by connecting on the AMI but that
doesn't seem to be doing anything either. So this channel is sitting there
in the 'BYE' state.
Is there anyway of clearing them without having to reload/restart Asterisk?
I want to see if that's the cause of the CPU usage and I'll lose that if I
restart Asterisk.
Thanks



> 2011/7/5, A E [Gmail] :
> > hello people,
> >
> > I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
> some
> > reason I have noticed that only after a few test calls, the asterisk
> process
> > is running between 95% - 99.9% CPU when there's absolutely nothing on the
> > system. This is a clean Asterisk system in an internal network with
> nothing
> > else on it with no calls on it but it's still sitting with 96% CPU.
> >
> > I'm not a developer so not that ept with using debug tools etc to figure
> out
> > why it's doing that. Could anyone please tell me how I can figure out why
> > it's doing this and/or help debug this. Makes no sense for it to be using
> > CPU with nothing happening on the system
> >
> > Thanks
> >
>
> --
> Enviado desde mi dispositivo móvil
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[Daniel - Asterisk]

On the CLI write: sip show channels

If there are lots of bye channels you have the same problem than me.
I've tried waiting with the call generator -sipp- and channels
finished when there are a few. But they're not ending faster enough
when I send lots of concurrent calls.

Elder

2011/7/5, A E [Gmail] :
> hello people,
>
> I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
> reason I have noticed that only after a few test calls, the asterisk process
> is running between 95% - 99.9% CPU when there's absolutely nothing on the
> system. This is a clean Asterisk system in an internal network with nothing
> else on it with no calls on it but it's still sitting with 96% CPU.
>
> I'm not a developer so not that ept with using debug tools etc to figure out
> why it's doing that. Could anyone please tell me how I can figure out why
> it's doing this and/or help debug this. Makes no sense for it to be using
> CPU with nothing happening on the system
>
> Thanks
>

-- 
Enviado desde mi dispositivo móvil

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[asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

hello people,

I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
reason I have noticed that only after a few test calls, the asterisk process
is running between 95% - 99.9% CPU when there's absolutely nothing on the
system. This is a clean Asterisk system in an internal network with nothing
else on it with no calls on it but it's still sitting with 96% CPU.

I'm not a developer so not that ept with using debug tools etc to figure out
why it's doing that. Could anyone please tell me how I can figure out why
it's doing this and/or help debug this. Makes no sense for it to be using
CPU with nothing happening on the system

Thanks
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[asterisk-users] Couldn't call Agent and segfault

[Agustina Berretta]

Hi folks!

I´m having the following problem:

I get the following messages, asterisk get automatically reloaded and agents
log out once or twice a day, randomly.

[Jul 4 11:36:25] VERBOSE[30004] app_queue.c: -- Couldn't call Agent/2002
[Jul 4 11:36:29] VERBOSE[30320] logger.c: Asterisk Event Logger Started
/var/log/asterisk/event_log
[Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing
'/etc/asterisk/asterisk.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: ==
Found
[Jul 4 11:36:29] VERBOSE[30320] loader.c: Asterisk Dynamic Loader Starting:
[Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing
'/etc/asterisk/modules.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: ==
Found
[Jul 4 11:36:29] NOTICE[30320] loader.c: 2 modules will be loaded.

Also I get a segfault in /var/log/messages.
Any help will be appreciated!
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Re: [asterisk-users] Asterisk 1.6.1 Realtime SIP Users

[Mickael MONSIEUR]

Thank you I'll watch. Support for Asterisk-Mysql is a bit minimal ... :-(

2011/7/1 Mickael MONSIEUR 

> Hello,
> I just implement the SIP Peers with MySQL.
>
> In the structure mySQL missing the following fields:
>
> nat = yes
> notransfer = yes
> dtmfmode = rfc2833
> call-limit = 2
> canreinvite = no
> subscribecontext = blf
>
> subscribecontext (BLF) and call-limit (Protection) are very important ...
> Can you help me?
>
> Best,
> Mickael
>
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[asterisk-users] OT - Polycom - 2 localization file versions on the same TFTP server

[Olivier]

Hi,

Using Polycom's Master configuration file, I could not find any convenient
way to store 2 different versions of the same localization file on the same
TFTP server.
Did I miss something ?

What I would like is to have both files under TFTP root
versionA/SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml
versionB/SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml

and tune [MAC_ADDRESS].cfg files so that each phone appropriately download
relevant SoundPointIP-dictionary.xml file.


I could find a hack creating my own fake language (for instance
French_France_FooBar) but it seems rather complex and it may break with
future upgrades.
Another hack is to create another TFTP server on the same machine.

Regards
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Re: [asterisk-users] realm question: solved

[Hans Witvliet]

On Tue, 2011-07-05 at 12:33 +0500, Faisal Hanif wrote:
> The problem you are reporting is not related to realm but can be context or
> domain.
> 
Tnx,
It was indeed a domain issue.
In some cases static definitions in /etc/hosts is not a good replacement
for DNS...

hw

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[asterisk-users] chanspy spies on wrong channel

[steve casto]

The argument to chanspy is a pattern and not an exact match.
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On Jul 2, 2011, at 3:48 PM, steve casto wrote:

> asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
> flash operator panel < 2.0
>
> (from extensions.conf)
> exten=> 304,1,ChanSpy(Zap/4|q)
> exten=> 304,2,hangup
> There is no entry ChanSpy(Zap/41)  in extensions.conf
>
> On dialing 304 and Zap/41 is in use this happens:
> [Jul  1 18:24:47] VERBOSE[14447] logger.c: -- Executing
> [304 at flash:1] ChanSpy("Zap/31-1", "Zap/4|q") in new stack
> [Jul  1 18:24:47] VERBOSE[14447] logger.c:   == Spying on channel Zap/41-1
> [Jul  1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to
> Zap/41-1
>
> If while spying on Zap/41 that channel is hung up:
> [Jul  1 19:06:48] VERBOSE[15242] logger.c:   == Done Spying on channel
> Zap/41-1
> [Jul  1 19:06:48] VERBOSE[15242] logger.c:   == Spying on channel Zap/4-1
> [Jul  1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to
Zap/4-1
>
> thanks list
> Steve
>
Was not aware of that. Using ExtenSpy now, works fine.
thanks
Steve


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Re: [asterisk-users] Blind Transfer Connected

[Kevin P. Fleming]

On 07/05/2011 01:54 AM, Olivier wrote:



2011/7/5 Nikhil mailto:d.nik...@cem-solutions.net>>

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

For Eg:
A make call to B through asterisk,then B transfer the call to C.
If C did not answer the call ,A  and B Call should connect back.

IMHO, blind tranfer definition is to NOT connect A and B back


That is correct, and is why it's called a 'blind' transfer; the 
transferring party does not know or care what happens to the call after 
effecting the transfer.


--
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Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Recording SIP history

[Lee Archer]

Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files.  When I sip show history
 it's fine but it's not logging anything.  My logger.conf has
debug => debug and the debug file grows.  Is my understanding correct in
that at the end of the call the entire sip show history  should
be dumped to the debug file?  I am using 1.6.2.19.

;--- SIP DEBUGGING
---
sipdebug=yes ; Turn on SIP debugging by default, from
; the moment the channel loads this
configuration
recordhistory=yes  ; Record SIP history by default
; (see sip history / sip no history)
dumphistory=yes; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG
logging channel

Thanks

Lee
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[asterisk-users] Can't get video on one server of 4

[Administrator TOOTAI]

Hi,

we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One 
GrandStream GXV3000 is used for the tests. He is registered to asterisk 
1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, 
get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP 
trunk from both others servers is also working well.


What fail, is video on echo test from asterisk 1.4.42 using SIP trunks: 
we have audio but no videobeside the fact that video codec are 
negociated as shown below.


All servers are on public IP. Here is a debug from a call from server 
running 1.4.35 asterisk to the 1.4.42 one:


<->
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: --- (14 headers 18 lines) 
---
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Sending to 
XXX.XXX.XXX.XXX : 5060 (no NAT)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Using INVITE request as 
basis request - 78938c042d374b341c4f1b60071d3...@xxx.xxx.xxx.xxx

[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found peer 'mypeer'
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 0
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 3
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 101
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description 
format PCMU for ID 0
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description 
format GSM for ID 3
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description 
format telephone-event for ID 101

[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 34
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 103
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 99
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description 
format H263 for ID 34
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description 
format h263-1998 for ID 103
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description 
format H264 for ID 99
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Capabilities: us - 
0x3c0002 (gsm|h261|h263|h263p|h264), peer - audio=0x380006 
(gsm|ulaw|h263|h263p|h264)/video=0x38 (h263|h263p|h264), combined - 
0x380002 (gsm|h263|h263p|h264)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Non-codec capabilities 
(dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), 
combined - 0x1 (telephone-event)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer audio RTP is at port 
XXX.XXX.XXX.XXX:40428
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer video RTP is at port 
XXX.XXX.XXX.XXX:44636
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Looking for 3800 in 
acces_groupe (domain mydomain.com)
[2011-07-05 16:08:14] VERBOSE[11535] logger.c: list_route: hop: 


[2011-07-05 16:08:14] VERBOSE[11535] logger.c:
<--- Transmitting (NAT) to XXX.XXX.XXX.XXX:5060 --->

All trunks. are setted from the same manier:

[trunk]
;
type=peer
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.XXX.XXX
host=host.domain.com
context=from-trunk
disallow=all
allow=all

A "sip show peer " show that video is on.

What can be the problem, I start loose my hairs!

Thanks for any hint.

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Re: [asterisk-users] Cant find asterisk src dir for FreePBX full distro

[Paul Belanger]

On 11-07-05 10:07 AM, Tobias Steen wrote:

It seems that the full distro package from FreePBX with Asterisk 1.8.1.4
someway hides (deletes?) the source directory for asterisk after
installation.

I cant find the directory under /usr/src/

I am trying to compile and install the conference module "app_konference"
and need to point a variable in Makefile to the src-dir of asterisk.

I think that FreePBX creates a user without root-access (named asterisk) and
installs the asterisk-distro with that user, is this correct? Can that be
the reason for not finding "/usr/src/asterisk" when logged in as root?


Does -devel provide it?

# yum install asterisk18-devel ?

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[asterisk-users] Cant find asterisk src dir for FreePBX full distro

[Tobias Steen]

It seems that the full distro package from FreePBX with Asterisk 1.8.1.4
someway hides (deletes?) the source directory for asterisk after
installation.

 

I cant find the directory under /usr/src/

 

I am trying to compile and install the conference module "app_konference"
and need to point a variable in Makefile to the src-dir of asterisk.

 

I think that FreePBX creates a user without root-access (named asterisk) and
installs the asterisk-distro with that user, is this correct? Can that be
the reason for not finding "/usr/src/asterisk" when logged in as root?





Best regards
Tobias

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[asterisk-users] AST_DEVICE_UNAVAILABLE vs. AST_DEVICE_UNKNOWN for new loaded realtime peers

[Kristijan Vrban]

Hello is use realtime sip-peers in 1.8, and have the problem, that when a peer
is loaded from database, the devstate is "AST_DEVICE_UNAVAILABLE" and
the the peers
can not be called from the queue. because the app_queue only calls
agens in state
AST_DEVICE_NOT_INUSE or AST_DEVICE_UNKNOWN.

My question: is this behavior configurable, or is s scource code
change necessary?
My opinion is, that when a sip peer is loaded from database, and it is
not a dynamic
host, the the devstate should be AST_DEVICE_UNKNOWN, because in this
moment it is
UNKNOWN, and the app_queue whould try to call the peer.

Kristijan

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Re: [asterisk-users] Load Balance Trunks

[Faisal Hanif]

Hi,

One of my college "Gohar Ahmed" suggested an intelligent solution to your
problem. I am coping his words below,

Create SIP trunks and create a queue [distributor] and register trunks in it
as static agents with strategy "rrmemory" , 
To keep track of number of calls served per trunk as well as time on each
trunk can be monitored via any queue monitoring tool. !!
 or better use queue_log in realtime DB

As per my view this is most easy and optimized approach while keeping all
possible data in queue logs. Hope this will helpful for you.

Regards,

Faisal Hanif

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, July 02, 2011 1:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Load Balance Trunks

On Fri, 1 Jul 2011, Abid Saleem wrote:

> The intention is to load balance between 100 or even more trunks. 
> Filling up one trunk may have another problem because we have another 
> restriction on 5 simultaneous calls per trunk. Yes unused capacity can 
> be rolled over to the next day. Anything is fine that does not break 
> these two restrictions of 120 mins/day/trunk and 5 simultaneous 
> calls/trunk.

> Please help me in writing an AGI script or whatever required if you 
> can as I am not a programmer.

If you don't consider yourself a 'programmer' then you don't have the skills
to start. You should hire a competent programmer. It will be much cheaper in
the long run and you can focus on what you are good at instead of what you
are not.

It's not that the requirements are all that challenging, it's just that the
probability of success when you lack the skills is small.

These skills include, but are not limited to:

1) An understanding of Asterisk, dialplan logic, and applications.

2) An understanding of the AGI interface including reading and setting
channel variables.

3) MySQL programming and administration skills.

4) The ability and experience to write well thought out, clearly presented,
robust and maintainable code.

What service are you offering?

Are the calls delivered by SIP or PSTN?

Is this a 24x7 operation?

If I was asked to design a 500 simultaneous call system with SIP delivery I
would probably start with 2 OpenSIPS servers, 2 Asterisk instances (possibly
on the same servers as the OpenSIPS servers), and at least 1 MySQL server.

You could cram everything on to a single system, I just don't like to put
all my eggs in a single basket.

I like 'front-ending' Asterisk servers with OpenSIPS because it gives me the
flexibility to handle a host failure or take a host out of production for
maintenance.

AJS (previous poster) has the right approach -- 2 AGIs. One AGI to determine
which trunk to use (I would use a 'select' to determine which trunk instead
of 'random') and one executed at the end of the call to update the database.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] SIP Presence not working

[Deka, Rajib IN MAA SL]

Hi All,

Following message I got in console for an extension,

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

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

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





Not online

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





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

Hello all,

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

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

Regards,
Rajib


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Re: [asterisk-users] More SQL Querys in dialplan

[Ulrich Meckel]

Everything was fine but i have a hangup in an other shorter route where 
the extenension was 1234, but i forgot the '4' so everytime it goes to 
the defined hangup in the other route.


Thx for your quick answer and sorry for my mistake


On 05.07.2011 11:34, Thorsten Göllner wrote:

Executing the query in MySQL-CLI is fine?

Am 05.07.2011 11:25, schrieb Ulrich Meckel:

Hi List

I tried to use SQL Query in my diaplan. If i only use one or two 
there is no Problem but if i try to start the third one after the 
other it hangup after the 2nd clear


exten => _123.,1,MYSQL(Connect connid host user  pw db_name)

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value1 FROM 
table1 WHERE sp=${var})

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var1)
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value2 FROM 
table2 WHERE sp='name')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var2)
exten => _123.,n,NoOp(var2 = ${var2})
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value3 FROM 
table2 WHERE sp='loc')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var3)
exten => _123.,n,NoOp(var3 = ${var3})
exten => _123.,n,MYSQL(Clear ${resultid}

exten => _123.,n,MYSQL(Disconnect ${connid})

CLI :
   -- Executing [03514138123@btsctrl:6] MYSQL("lcr/5", "Fetch fetchid 
2 var2") in new stack
   -- Executing [03514138123@btsctrl:7] NoOp("lcr/5", "var2= 49") in 
new stack
   -- Executing [03514138123@btsctrl:8] MYSQL("lcr/5", "Clear 2") in 
new stack
   -- Executing [03514138123@btsctrl:9] Hangup("lcr/5", "") in new stack 



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Fröbelstr. 57, 01159 Dresden  Fax:   0351/41381 - 12
___

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USt.Id DE211326547  Mail:  netzing...@netzing.de


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Re: [asterisk-users] More SQL Querys in dialplan

[Thorsten Göllner]

Executing the query in MySQL-CLI is fine?

Am 05.07.2011 11:25, schrieb Ulrich Meckel:

Hi List

I tried to use SQL Query in my diaplan. If i only use one or two there 
is no Problem but if i try to start the third one after the other it 
hangup after the 2nd clear


exten => _123.,1,MYSQL(Connect connid host user  pw db_name)

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value1 FROM 
table1 WHERE sp=${var})

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var1)
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value2 FROM 
table2 WHERE sp='name')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var2)
exten => _123.,n,NoOp(var2 = ${var2})
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value3 FROM 
table2 WHERE sp='loc')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var3)
exten => _123.,n,NoOp(var3 = ${var3})
exten => _123.,n,MYSQL(Clear ${resultid}

exten => _123.,n,MYSQL(Disconnect ${connid})

CLI :
   -- Executing [03514138123@btsctrl:6] MYSQL("lcr/5", "Fetch fetchid 
2 var2") in new stack
   -- Executing [03514138123@btsctrl:7] NoOp("lcr/5", "var2= 49") in 
new stack
   -- Executing [03514138123@btsctrl:8] MYSQL("lcr/5", "Clear 2") in 
new stack
   -- Executing [03514138123@btsctrl:9] Hangup("lcr/5", "") in new stack 



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[asterisk-users] More SQL Querys in dialplan

[Ulrich Meckel]

Hi List

I tried to use SQL Query in my diaplan. If i only use one or two there 
is no Problem but if i try to start the third one after the other it 
hangup after the 2nd clear


exten => _123.,1,MYSQL(Connect connid host user  pw db_name)

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value1 FROM 
table1 WHERE sp=${var})

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var1)
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value2 FROM 
table2 WHERE sp='name')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var2)
exten => _123.,n,NoOp(var2 = ${var2})
exten => _123.,n,MYSQL(Clear ${resultid})

exten => _123.,n,MYSQL(Query resultid ${connid} SELECT value3 FROM 
table2 WHERE sp='loc')

exten => _123.,n,MYSQL(Fetch fetchid ${resultid} var3)
exten => _123.,n,NoOp(var3 = ${var3})
exten => _123.,n,MYSQL(Clear ${resultid}

exten => _123.,n,MYSQL(Disconnect ${connid})

CLI :
   -- Executing [03514138123@btsctrl:6] MYSQL("lcr/5", "Fetch fetchid 2 
var2") in new stack
   -- Executing [03514138123@btsctrl:7] NoOp("lcr/5", "var2= 49") in 
new stack
   -- Executing [03514138123@btsctrl:8] MYSQL("lcr/5", "Clear 2") in 
new stack

   -- Executing [03514138123@btsctrl:9] Hangup("lcr/5", "") in new stack


Thank you in advance for your help

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Re: [asterisk-users] realm question

[Faisal Hanif]

The problem you are reporting is not related to realm but can be context or
domain.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Tuesday, July 05, 2011 11:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: j.witvl...@mindef.nl
Subject: [asterisk-users] realm question

Hi all,

Trying to find where i got wrong in my config

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

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

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

Kind regards, Hans


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Re: [asterisk-users] stream rtp from asterisk

[Marcus Kvarsell]

Yes, i have done this already. Though there is no possibility of sending unique 
id or just recording answered calls with the oreka GPL version. This is where 
the xorcom asterisk patch comes in handy, because you can set it to start 
sending the trp data when a call gets into the queue.

/ Marcus

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För Johan Wilfer
Skickat: den 4 juli 2011 16:56
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] stream rtp from asterisk

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

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

/Johan
>
>
>
>
> -Ursprungligt meddelande-
> Från: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov
> Skickat: den 4 juli 2011 14:49
> Till: asterisk-users@lists.digium.com
> Ämne: Re: [asterisk-users] stream rtp from asterisk
>
> On 07/04/2011 06:58 AM, Marcus Kvarsell wrote:
>
>> Anybody familiar with streaming rtp from asterisk. Preferably with 
>> the xorcom asterisk patch which streams rtp from asterisk to oreka 
>> audio server. Any ideas will do just fine though!
> Can you clarify what you mean by "streaming"?
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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--
Med vänlig hälsning

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


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