[asterisk-users] asterisk and installing chan_h323.so rpm

Bincy K. Philip bincy.philip at nestgroup.net
Wed Nov 14 05:47:08 CST 2007 


Hello,

 While trying to install H323 support for asterisk, I missed one step.
After compiling files in channel/h323, need to select chanh323 from the menu and compile and install asterisk.


cd asterisk

*  ./configure

 * make menuconfig

 channel drivers->chanh323

 save the setting by giving x.


*  make 
   make

* make install



Hope this will help someone.



Thanks & Regards
Bincy K Philip


-----Original Message-----
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Sent: Tuesday, November 13, 2007 2:56 PM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 40, Issue 34


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Today's Topics:

   1. Re: asterisk and  installing chan_h323.so rpm (Dovid B)
   2. Re: Asterisk direct dialing (Dovid B)
   3. Re: Help: Static and dropped calls (Dovid B)
   4. Re: How to delete voice mail messages? (Dovid B)
   5. Re: asterisk and  installing chan_h323.so rpm (David Boyd)
   6. Re: ztdummy and BackGround (Tony Plack)
   7. MOH Codec Issue (Nick Brown)
   8. Re: RTP traffic not being forwarded (Ryan Newington)
   9. Re: MOH Codec Issue (Paul Hales)
  10. Chatterbug (Robert Goodyear)
  11. Re: MOH Codec Issue (Nick Brown)
  12. Re: Chatterbug (Paul Hales)
  13. Re: MOH Codec Issue (Paul Hales)
  14. Re: MOH Codec Issue (Paul Hales)
  15. Stress-Testing Asterisk (Jeng Yu)
  16. Re: Stress-Testing Asterisk (Suity Zsolt)
  17. Re: Stress-Testing Asterisk (Tzafrir Cohen)
  18. Toshiba DK - Asterisk Integration (Indika Wasala)
  19. Fwd: Re:  Grandstream GXP2020 + Asterisk 1.4.11 (Erik Wartusch)


----------------------------------------------------------------------

Message: 1
Date: Tue, 13 Nov 2007 03:00:22 +0200
From: "Dovid B" <asteriskusers at dovid.net>
Subject: Re: [asterisk-users] asterisk and  installing chan_h323.so
	rpm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <023901c82590$c7f2b6f0$6402a8c0 at DovidLaptop>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
	reply-type=original


----- Original Message ----- 
From: "Bincy K. Philip" <bincy.philip at nestgroup.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, November 08, 2007 2:13 PM
Subject: [asterisk-users] asterisk and installing chan_h323.so rpm


> Hello,
>
> When I tried to install chan_h323-1.0.1-module.i386 RPM i got these 
> errors.
>
>
> Failed dependencies:
>        libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>   libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>
> But i found the same files in
>
> /usr/lib/libh323_linux_x86_r.so.1
> /usr/lib/libpt_linux_x86_r.so.1
>
>
> What to do for asterisk to detect the same files?
>
I had this issue in the past. I do not remember what I did to resolve it. In 
the end I went with the h323 channel driver located in the asterisk add-ons. 
It was a lot easier to work with and worked with out any issues. 





------------------------------

Message: 2
Date: Tue, 13 Nov 2007 03:02:08 +0200
From: "Dovid B" <asteriskusers at dovid.net>
Subject: Re: [asterisk-users] Asterisk direct dialing
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <023f01c82590$d32e14b0$6402a8c0 at DovidLaptop>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
	reply-type=original


----- Original Message ----- 
From: "Gopal krishnan" <saigop at gmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, November 10, 2007 12:46 PM
Subject: [asterisk-users] Asterisk direct dialing


> Hi,
>     I am using Asterisk 1.2.24, I have written my dialplan to land
> with an IVR with the same time if the customer knows the parties
> extensions they can dial directly, but what happens is sometimes its
> working and sometime its not working.
> My extensions.conf as follows,
>
> [incoming]
> exten => 052477302,1,Wait(2)
> exten => 052477302,2,NoOp(${CALLERIDNUM})
> exten => 052477302,3,Goto(from-internal,s,1)
>
> [from-internal]
> exten => s,1,Answer()
> exten => s,2,Background(welcome_pride5)
> exten => 501,1,Dial(Zap/1)
> exten => 502,1,Dial(Zap/2)
> exten => 503,1,Dial(Zap/3)
> exten => 504,1,Dial(Zap/4)
> exten => 505,1,Dial(Zap/5)
>
> I dont know what could be the reason. Is there any other way that i can 
> use.
>
> -- 
> Thank you, with regards,
> Gopal,
>

What comes up in the CLI when it does not work ? 





------------------------------

Message: 3
Date: Tue, 13 Nov 2007 03:10:36 +0200
From: "Dovid B" <asteriskusers at dovid.net>
Subject: Re: [asterisk-users] Help: Static and dropped calls
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <039801c82592$0194a1b0$6402a8c0 at DovidLaptop>
Content-Type: text/plain; charset="iso-8859-1"

You need to provide more information that just that. Maybe a CLI output ? Have you tested with any other providers ? SIP debug ? Ran a trace ? We aren't mind readers here. 
  ----- Original Message ----- 
  From: Jarga Jallow 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Monday, November 05, 2007 8:27 PM
  Subject: [asterisk-users] Help: Static and dropped calls


   
  Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters.

   

  Thank you

   

  Jarga Jallow



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Message: 4
Date: Tue, 13 Nov 2007 03:14:48 +0200
From: "Dovid B" <asteriskusers at dovid.net>
Subject: Re: [asterisk-users] How to delete voice mail messages?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <03e301c82592$97e30670$6402a8c0 at DovidLaptop>
Content-Type: text/plain; charset="iso-8859-1"

I wrote this for a client a while back:

[del-all-vm]
exten => s,1,Set(TIMEOUT(digit)=3)
exten => s,2,Set(TIMEOUT(response)=6)
exten => s,3,Background(enter-exten-for-vm-to-delete)
exten => _XX,1,Set(THIER_EXTEN=${EXTEN})
exten => _XX,2,Goto(del-all-vm-confirm,s,1)
exten => i,1,Playback(invalid)
exten => i,2,Goto(s,1)
exten => t,1,Goto(s,1)

[del-all-vm-confirm]
exten => s,1,Set(TIMEOUT(digit)=3)
exten => s,2,Set(TIMEOUT(response)=6)
exten => s,3,Background(are-you-sure-del-all-vm)
exten => s,4,Saynumber(${THIER_EXTEN})
exten => s,5,Background(1-for-yes-2-for-no)
exten => 1,1,System(rm -rf /var/spool/asterisk/default/techmast/${THIER_EXTEN}/INBOX/*.*)
exten => 1,2,Playback(all-vm-deleted)
exten => 1,3,Congestion
exten => 1,4,Hangup
exten => 2,1,Playback(close-call-not-deleted)
exten => 2,2,Congestion
exten => 2,3,Hangup
exten => i,1,Playback(invalid)
exten => i,2,Goto(s,1)
exten => t,1,Goto(s,1)
  ----- Original Message ----- 
  From: voip crazy 
  To: asterisk-users at lists.digium.com 
  Sent: Monday, November 05, 2007 1:15 PM
  Subject: [asterisk-users] How to delete voice mail messages?


  Hello all,

  Could I create a script to delete the first messages on my voice mail? In this script should I update any "messages index file" or there isn't any file  to index them? Could you share any script to do that? 

  Thanks in advance.

  VoipCrazy. 



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------------------------------

Message: 5
Date: Mon, 12 Nov 2007 20:30:25 -0500 (EST)
From: "David Boyd" <dboyd at ignitetrx.com>
Subject: Re: [asterisk-users] asterisk and  installing chan_h323.so
	rpm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<3836.71.62.178.191.1194917425.squirrel at www.gordonramsey.org>
Content-Type: text/plain;charset=iso-8859-1

>
> ----- Original Message -----
> From: "Bincy K. Philip" <bincy.philip at nestgroup.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, November 08, 2007 2:13 PM
> Subject: [asterisk-users] asterisk and installing chan_h323.so rpm
>
>
>> Hello,
>>
>> When I tried to install chan_h323-1.0.1-module.i386 RPM i got these
>> errors.
>>
>>
>> Failed dependencies:
>>        libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>>   libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>>
>> But i found the same files in
>>
>> /usr/lib/libh323_linux_x86_r.so.1
>> /usr/lib/libpt_linux_x86_r.so.1
>>
>>
>> What to do for asterisk to detect the same files?
>>
> I had this issue in the past. I do not remember what I did to resolve it.
> In
> the end I went with the h323 channel driver located in the asterisk
> add-ons.
> It was a lot easier to work with and worked with out any issues.
>


It seems to me that you need to run ldconfig so as to pick up the location
of the specified libraries.  Do a google on it to see syntax of "man
ldconfig".

You could also hack things by linking to the libraries from the expected
directories (What the rpm is expecting) if executing ldconfig doesn't
work.

db

>




------------------------------

Message: 6
Date: Mon, 12 Nov 2007 20:25:49 -0600
From: Tony Plack <Tony at plack.net>
Subject: Re: [asterisk-users] ztdummy and BackGround
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <20071112202549.444788 at NeuralMonk>
Content-Type: text/plain; charset="us-ascii"

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------------------------------

Message: 7
Date: Tue, 13 Nov 2007 15:04:56 +1100
From: Nick Brown <Nick at ipera.com.au>
Subject: [asterisk-users] MOH Codec Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <C35F6D98.A3B%Nick at ipera.com.au>
Content-Type: text/plain; charset="us-ascii"

Afternoon All,

Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to
get a working SCCP channel). During the process Music On Hold appears to
have died (Not, just when calling from a SCCP device, but coming in on SIP
also).

CLI is showing

    -- Executing [XXXXXXXX at unauthed-inbound:2]
MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
[Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a
codec translation path from alaw to unknown
[Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to
set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
    -- Started music on hold, class '?S?', on channel
'SIP/10.97.1.33-09f0cfc8'
[Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to
start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8

Have attempted to use an alternate Music On Hold context and forced a
format= within musiconhold.conf.

Otherwise all other audio (Playback, voice etc) seems fine.

Anyone seen this before? Can not see anything in the tracker regarding this
issue in 1.4.7 specifically.

Cheers
Nick.
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Message: 8
Date: Tue, 13 Nov 2007 15:26:52 +1100
From: Ryan Newington <ryan at lithiumblue.com>
Subject: Re: [asterisk-users] RTP traffic not being forwarded
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<E36FF4223E93DD45B2342102343080CB453B715A at dc2.lithnet.local>
Content-Type: text/plain; charset="us-ascii"

Hi Vivek,

Thanks for the link. I had a look through and couldn't find anything that worked. There are no NAT problems as this is all taking place on my internal network. The rtp.conf is used to configure the ports. There are no firewalls or gateways in between these devices.

Asterisk is listening on the correct ports, and receiving the traffic, as no ICMP messages are being generated to say that the packets could not be delivered.

Ryan


From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded

well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate.

http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html

Let me know if you need more information.

Thanks,

Vivek



On 11/11/07, Ryan Newington <ryan at lithiumblue.com<mailto:ryan at lithiumblue.com>> wrote:



Hi Vivek,



I'm not sure what you mean, could you explain further?



Regards



Ryan





From: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 1:21 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded



Hi Ryan,



I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf)



Regards,



Vivek



On 11/11/07, Ryan Newington <ryan at lithiumblue.com<mailto:ryan at lithiumblue.com>> wrote:

Hi Vivek,



The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000.

I can change the port range on the media server, asterisk and the device, but neither seems to help.



My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk)



SIP Phone <---> Media Gateway ---> Asterisk <--- SIP Phone



Ryan





From: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Vivek Shrivastava
Sent: Sunday, 11 November 2007 5:19 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded



Hi Ryan,



Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking

random port selection option on the device/softphone may help.



--Vivek



On 11/10/07, Ryan Newington <ryan at lithiumblue.com<mailto:ryan at lithiumblue.com>> wrote:

Hi Luki,

Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success.

Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway.

SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone

An asterisk internal call will work fine. Eg;

SIP Phone <-> Asterisk <-> SIP Phone

Regards

Ryan



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> ] On Behalf Of Luki
Sent: Sunday, 11 November 2007 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded

> When using 'rtp debug' on the asterisk console, it shows that it is
> receiving traffic from one endpoint, but not the other. A wireshark trace
> reveals it is actually receiving traffic from both ends.

Sounds like a firewall issue. Wireshark shows what's "on the wire",
i.e. before iptables. The packets are being dropped for whatever
reason and never reach the asterisk process. Check your iptables and
RTP port range, and perhaps try changing it.

Luki

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------------------------------

Message: 9
Date: Tue, 13 Nov 2007 15:56:21 +1100
From: Paul Hales <pdhales at optusnet.com.au>
Subject: Re: [asterisk-users] MOH Codec Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Cc: Nick Brown <Nick at ipera.com.au>
Message-ID: <1194929781.14827.37.camel at localhost.localdomain>
Content-Type: text/plain


What format is your music on hold in? 

PaulH


On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
> Afternoon All,
> 
> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
> attempt to get a working SCCP channel). During the process Music On
> Hold appears to have died (Not, just when calling from a SCCP device,
> but coming in on SIP also).
> 
> CLI is showing
> 
>     -- Executing [XXXXXXXX at unauthed-inbound:2]
> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to
> find a codec translation path from alaw to unknown
> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc:
> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
>     -- Started music on hold, class '?S?', on channel
> 'SIP/10.97.1.33-09f0cfc8'
> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec:
> Unable to start music on hold (class 'sounds') on channel
> SIP/10.97.1.33-09f0cfc8
> 
> Have attempted to use an alternate Music On Hold context and forced a
> format= within musiconhold.conf.
> 
> Otherwise all other audio (Playback, voice etc) seems fine.
> 
> Anyone seen this before? Can not see anything in the tracker regarding
> this issue in 1.4.7 specifically.
> 
> Cheers
> Nick. 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




------------------------------

Message: 10
Date: Mon, 12 Nov 2007 21:07:11 -0800
From: Robert Goodyear <me at jrob.net>
Subject: [asterisk-users] Chatterbug
To: asterisk-users at lists.digium.com
Message-ID: <F919BB80-D182-4B12-9532-69FA037913BA at jrob.net>
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes

Does anyone know anything about the Chatterbug product? I can't tell  
if it's an ATA with a modem or some sort of LCR proxy or somesuch.

Anyone?




------------------------------

Message: 11
Date: Tue, 13 Nov 2007 16:20:42 +1100
From: Nick Brown <Nick at ipera.com.au>
Subject: Re: [asterisk-users] MOH Codec Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <C35F7F5A.A43%Nick at ipera.com.au>
Content-Type: text/plain;	charset="ISO-8859-1"

It was using the 3 wav's from Freeplay. I have just recompiled and told it
to pull down the ULAW versions, then removed the Wav's however it has made
no difference.

Cheers
Nick

On 13/11/07 3:56 PM, "Paul Hales" wrote:

> 
> What format is your music on hold in?
> 
> PaulH
> 
> 
> On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
>> Afternoon All,
>> 
>> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
>> attempt to get a working SCCP channel). During the process Music On
>> Hold appears to have died (Not, just when calling from a SCCP device,
>> but coming in on SIP also).
>> 
>> CLI is showing
>> 
>>     -- Executing [XXXXXXXX at unauthed-inbound:2]
>> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
>> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to
>> find a codec translation path from alaw to unknown
>> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc:
>> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
>>     -- Started music on hold, class '?S?', on channel
>> 'SIP/10.97.1.33-09f0cfc8'
>> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec:
>> Unable to start music on hold (class 'sounds') on channel
>> SIP/10.97.1.33-09f0cfc8
>> 
>> Have attempted to use an alternate Music On Hold context and forced a
>> format= within musiconhold.conf.
>> 
>> Otherwise all other audio (Playback, voice etc) seems fine.
>> 
>> Anyone seen this before? Can not see anything in the tracker regarding
>> this issue in 1.4.7 specifically.
>> 
>> Cheers
>> Nick. 
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

Regards,
Nick Brown

Ipera Communications Pty Ltd
Level 1, 9 Denison Street,?????????????
Newcastle West NSW 2302
PO Box 2115, Dangar NSW 2309???????

? P: +61 2 4910 1000
? F: +61 2 4910 1099
? ABN: 31 090 964 104




------------------------------

Message: 12
Date: Tue, 13 Nov 2007 16:21:07 +1100
From: Paul Hales <pdhales at optusnet.com.au>
Subject: Re: [asterisk-users] Chatterbug
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <1194931267.14827.43.camel at localhost.localdomain>
Content-Type: text/plain


http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf

PaulH


On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote:
> Does anyone know anything about the Chatterbug product? I can't tell  
> if it's an ATA with a modem or some sort of LCR proxy or somesuch.
> 
> Anyone?
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




------------------------------

Message: 13
Date: Tue, 13 Nov 2007 16:48:58 +1100
From: Paul Hales <pdhales at optusnet.com.au>
Subject: Re: [asterisk-users] MOH Codec Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <1194932938.14827.46.camel at localhost.localdomain>
Content-Type: text/plain; charset=utf-8


Is it possibly a funny zaptel issue?

Paul Hales
AsteriskIT


On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote:
> It was using the 3 wav's from Freeplay. I have just recompiled and told it
> to pull down the ULAW versions, then removed the Wav's however it has made
> no difference.
> 
> Cheers
> Nick
> 
> On 13/11/07 3:56 PM, "Paul Hales" wrote:
> 
> > 
> > What format is your music on hold in?
> > 
> > PaulH
> > 
> > 
> > On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
> >> Afternoon All,
> >> 
> >> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
> >> attempt to get a working SCCP channel). During the process Music On
> >> Hold appears to have died (Not, just when calling from a SCCP device,
> >> but coming in on SIP also).
> >> 
> >> CLI is showing
> >> 
> >>     -- Executing [XXXXXXXX at unauthed-inbound:2]
> >> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
> >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to
> >> find a codec translation path from alaw to unknown
> >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc:
> >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
> >>     -- Started music on hold, class '?S?', on channel
> >> 'SIP/10.97.1.33-09f0cfc8'
> >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec:
> >> Unable to start music on hold (class 'sounds') on channel
> >> SIP/10.97.1.33-09f0cfc8
> >> 
> >> Have attempted to use an alternate Music On Hold context and forced a
> >> format= within musiconhold.conf.
> >> 
> >> Otherwise all other audio (Playback, voice etc) seems fine.
> >> 
> >> Anyone seen this before? Can not see anything in the tracker regarding
> >> this issue in 1.4.7 specifically.
> >> 
> >> Cheers
> >> Nick. 
> >> _______________________________________________
> >> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >> 
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> Regards,
> Nick Brown
> 
> Ipera Communications Pty Ltd
> Level 1, 9 Denison Street,             
> Newcastle West NSW 2302
> PO Box 2115, Dangar NSW 2309       
> 
> ? P: +61 2 4910 1000
> ? F: +61 2 4910 1099
> ? ABN: 31 090 964 104
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 




------------------------------

Message: 14
Date: Tue, 13 Nov 2007 16:48:58 +1100
From: Paul Hales <pdhales at optusnet.com.au>
Subject: Re: [asterisk-users] MOH Codec Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <1194932938.14827.46.camel at localhost.localdomain>
Content-Type: text/plain; charset=utf-8


Is it possibly a funny zaptel issue?

Paul Hales
AsteriskIT


On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote:
> It was using the 3 wav's from Freeplay. I have just recompiled and told it
> to pull down the ULAW versions, then removed the Wav's however it has made
> no difference.
> 
> Cheers
> Nick
> 
> On 13/11/07 3:56 PM, "Paul Hales" wrote:
> 
> > 
> > What format is your music on hold in?
> > 
> > PaulH
> > 
> > 
> > On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
> >> Afternoon All,
> >> 
> >> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
> >> attempt to get a working SCCP channel). During the process Music On
> >> Hold appears to have died (Not, just when calling from a SCCP device,
> >> but coming in on SIP also).
> >> 
> >> CLI is showing
> >> 
> >>     -- Executing [XXXXXXXX at unauthed-inbound:2]
> >> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
> >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to
> >> find a codec translation path from alaw to unknown
> >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc:
> >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
> >>     -- Started music on hold, class '?S?', on channel
> >> 'SIP/10.97.1.33-09f0cfc8'
> >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec:
> >> Unable to start music on hold (class 'sounds') on channel
> >> SIP/10.97.1.33-09f0cfc8
> >> 
> >> Have attempted to use an alternate Music On Hold context and forced a
> >> format= within musiconhold.conf.
> >> 
> >> Otherwise all other audio (Playback, voice etc) seems fine.
> >> 
> >> Anyone seen this before? Can not see anything in the tracker regarding
> >> this issue in 1.4.7 specifically.
> >> 
> >> Cheers
> >> Nick. 
> >> _______________________________________________
> >> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >> 
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> Regards,
> Nick Brown
> 
> Ipera Communications Pty Ltd
> Level 1, 9 Denison Street,             
> Newcastle West NSW 2302
> PO Box 2115, Dangar NSW 2309       
> 
> ? P: +61 2 4910 1000
> ? F: +61 2 4910 1099
> ? ABN: 31 090 964 104
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 




------------------------------

Message: 15
Date: Tue, 13 Nov 2007 07:22:17 +0000 (GMT)
From: Jeng Yu <jengyu2007 at yahoo.co.uk>
Subject: [asterisk-users] Stress-Testing Asterisk
To: asterisk-users at lists.digium.com
Message-ID: <583063.75150.qm at web23002.mail.ird.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1

Hi All,

I was wondering, what tools are readily available out
there in Asteriskland for me to use in stress/load
testing asterisk box I have in the lab. I want to
observe how my box holds out under heavy/light/medium
load.

Thanks,

Jeng


      ___________________________________________________________ 
Want ideas for reducing your carbon footprint? Visit Yahoo! For Good  http://uk.promotions.yahoo.com/forgood/environment.html




------------------------------

Message: 16
Date: Tue, 13 Nov 2007 09:00:06 +0100
From: Suity Zsolt <suich at yunord.net>
Subject: Re: [asterisk-users] Stress-Testing Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <47395986.50903 at yunord.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Jeng Yu wrote:
> Hi All,
> 
> I was wondering, what tools are readily available out
> there in Asteriskland for me to use in stress/load
> testing asterisk box I have in the lab. I want to
> observe how my box holds out under heavy/light/medium
> load.

Try SIPp from HP (http://sipp.sourceforge.net/index.html)
or  SIP swiss army knife - SIPSAK (http://sipsak.org/)



--
Suich



------------------------------

Message: 17
Date: Tue, 13 Nov 2007 10:21:17 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] Stress-Testing Asterisk
To: asterisk-users at lists.digium.com
Message-ID: <20071113082117.GR6765 at xorcom.com>
Content-Type: text/plain; charset=us-ascii

On Tue, Nov 13, 2007 at 09:00:06AM +0100, Suity Zsolt wrote:
> Jeng Yu wrote:
> > Hi All,
> > 
> > I was wondering, what tools are readily available out
> > there in Asteriskland for me to use in stress/load
> > testing asterisk box I have in the lab. I want to
> > observe how my box holds out under heavy/light/medium
> > load.
> 
> Try SIPp from HP (http://sipp.sourceforge.net/index.html)
> or  SIP swiss army knife - SIPSAK (http://sipsak.org/)

and also:

Asterisk (http://asterisk.org/) is a highly programmable and
configurable PBX. Use one on a stornger box or several "client" boxes.
This can allow you to stress a box through SIP, IAX, H323, or whatever.

(/me wonders if NBS could be used to stress-test an Asterisk box :-)

-- 
               Tzafrir Cohen       
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com       
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir



------------------------------

Message: 18
Date: Tue, 13 Nov 2007 13:56:02 +0530
From: Indika Wasala <indika at hsenid.lk>
Subject: [asterisk-users] Toshiba DK - Asterisk Integration
To: asterisk-users at lists.digium.com
Message-ID: <47395F9A.7060302 at hsenid.lk>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi All,

I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 
separate offices as follows,

Toshiba Strata dk28
Toshiba Strata dk280
Toshiba Strata dk8

I need to install 3 Asterisk servers in these 3 locations and integrate 
them with each of the Toshiba PBX s. This is to give IP Phones/soft 
phones to the users and to route these VOIP calls through the PBX to 
POTS. What are the Digium cards I should use in each of these cases and 
How should I integrate Asterisk with above systems.

I read the article in 
http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and 
not sure whether that scenario fits mine. Also it is bit confusing to 
identify what Digium cards should I need for my cases.

Any help is highly appreciated.

Thanks,
Indika.



------------------------------

Message: 19
Date: Tue, 13 Nov 2007 10:25:33 +0100
From: Erik Wartusch <we at deuromedia.at>
Subject: [asterisk-users] Fwd: Re:  Grandstream GXP2020 + Asterisk
	1.4.11
To: asterisk-users at lists.digium.com
Message-ID: <200711131025.33580.we at deuromedia.at>
Content-Type: text/plain;  charset="iso-8859-1"


Thx John !!

Hmm I found now on voip-info.org a lot of Beta releases which should fix my 
problems... Kind of strange whats going on with Grandstream devices and their 
firmware ... If you install the latest "official" release you can expect a 
few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped 
calls).  So you have to install the BETAS whether you want or not... 

That you have to use unique ports is a rumour and not SIP standard. As John 
said --> IP:Port must be unique . I definitely not understand why I should 
use random ports.

Kind Regards,

Erik

____________

> I`m using several GXP2020 phones with newest Firmware 1.1.4.18.

I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22
and have eliminated that.

> Asterisk Version: 1.4.11.

Me too. Still testing 1.4.13 on a non-production system.

> I use on every phone the 10000 as local port and in the rtp.conf
>
>From my knowledge of IP I don't think this is a problem since the

address/port would be unique. However the example config I originally had
from Grandstream indicated that each phone should use a different port and
recommended to use the random port option on the phones. I have since
assigned the port number on each phone to 10000 plus the extension number.
This was done to create a unique port number and to help with
troubleshooting when using Wireshark or tcpdump. I set this in the config
file for each phone.

John


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