[asterisk-users] res_perl with asterisk 1.4 compile problem

2006-12-28 Thread Gentian Bajraktari 
Dear all,

now we have the same problem of res_perl compilation with asterisk 1.4. It is 
the same problem that was present when asterisk was upgraded to version 1.2. 

I hope Anthony Minessale will be able to solve that problem as he did on that 
case. But if any of you know a hack to this problem please let us know. 

Here is the same compile problem again: 

gcc -Wall -DRES_PERL_BASE=\/usr/local/res_perl\ -DMULTIPLICITY -D_REENTRANT 
-D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe 
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE 
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm 
-I/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE -I/usr/src/asterisk 
-I/usr/src/asterisk/include -I. -c res_perl.c 
In file included from ./res_perl.h:17, 
from res_perl.c:17: 
/usr/src/asterisk/include/asterisk/module.h:204: warning: struct ast_channel 
declared inside parameter list 
/usr/src/asterisk/include/asterisk/module.h:204: warning: its scope is only 
this definition or declaration, which is probably not what you want 
In file included from ./res_perl.h:22, 
from res_perl.c:17: 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: error: 
syntax error before perl_mutex 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: warning: 
type defaults to `int' in declaration of `perl_mutex' 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: warning: 
data definition has no type or storage class 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: error: 
syntax error before perl_cond 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: warning: 
type defaults to `int' in declaration of `perl_cond' 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: warning: 
data definition has no type or storage class 
In file included from 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:3988, 
from ./res_perl.h:22, 
from res_perl.c:17: 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: error: 
syntax error before PL_op_mutex 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: warning: 
type defaults to `int' in declaration of `PL_op_mutex' 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: warning: 
data definition has no type or storage class 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: error: 
syntax error before PL_dollarzero_mutex 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: warning: 
type defaults to `int' in declaration of `PL_dollarzero_mutex' 
/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: warning: 
data definition has no type or storage class 
res_perl.c:492: warning: initialization from incompatible pointer type 
res_perl.c: In function `_update_perl': 
res_perl.c:837: warning: ISO C90 forbids mixed declarations and code 
res_perl.c: At top level: 
res_perl.c:921: warning: initialization from incompatible pointer type 
res_perl.c:1023: warning: initialization from incompatible pointer type 
res_perl.c: In function `_load_module': 
res_perl.c:1036: warning: ISO C90 forbids mixed declarations and code 
make: *** res_perl.o Error 1 


Rg,

Gentian
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[Asterisk-Users] res_perl error when loading asterisk

2005-12-07 Thread Gentian Bajraktari 



Hi,

I have a problem with loading res_perl ins asterisk 
1.2 stable:


res_perl.so]Dec 5 03:06:53 WARNING[11563]: 
loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined 
symbol: ast_bridge_callDec 5 03:06:53 WARNING[11563]: loader.c:554 
load_modules: Loading module res_perl.so failed!

My specs:

I am using STABLE 1.2 from tar.gz package , 
FedoraCore 4, updated with yum, perl 5.8.7


Rg.
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Re: [Asterisk-Users] auto assigning SIP port

2005-11-18 Thread Gentian Bajraktari 



yes you can do that by using provisioning. everytime the ata 
restarts it will get a new port

  - Original Message - 
  From: 
  xcel 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, November 18, 2005 7:00 
  PM
  Subject: [Asterisk-Users] auto assigning 
  SIP port
  
  
  
  is there anyway i can assign sip ports to clients (e,g ATA device or 
  softphones) from server ??? 
  my service provider hav blocked all the SIP standard port and everytime i 
  have to manually configure the port at each cleint end.
  Im looking for a way where i can deifne this port globally on server and 
  this server should tell the client which port to use for sip.
  
  
  --xce
  
  

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Re: [Asterisk-Users] Provisioning server

2005-11-18 Thread Gentian Bajraktari 



yes you can do that.

  - Original Message - 
  From: 
  xcel 
  To: asterisk-users@lists.digium.com 
  ; asterisk-users@lists.digium.com 
  
  Sent: Friday, November 18, 2005 7:02 
  PM
  Subject: RE: [Asterisk-Users] 
  Provisioning server
  
  initially for Sipura SPA 1001.
  
  
  
  Regards,
  *** REPLY SEPARATOR 
  ***On 11/18/2005 at 12:44 PM Jonathan k. Creasy 
  wrote:
  

For which 
equipment? 






From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of xcelSent: Friday, November 18, 2005 11:53 
AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Provisioning 
server


Can any one help me in setting 
up Provisioning sever ??




  
  

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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-14 Thread Gentian Bajraktari 



Dependent on what Channel you are 
using..
If you are using SIP then do:

*CLI sip show channels


Peer 
User/ANR Call ID Seq 
(Tx/Rx) Format Last Msg



Where Format will show the current codecs used at 
that time..

Rg,

Gentian


  - Original Message - 
  From: 
  Mark 
  Quitoriano 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, November 14, 2005 12:13 
  PM
  Subject: Re: [Asterisk-Users] How to 
  check how many G729 codec licenseinstalled
  how can i check how many g729 are being used right now?
  On 11/13/05, Gentian 
  Bajraktari [EMAIL PROTECTED]  
  wrote:
  Yes.- 
Original Message -From: "Angelito Manansala"  [EMAIL PROTECTED]To: 
"Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.com 
Sent: Sunday, November 13, 2005 1:23 PMSubject: Re: 
[Asterisk-Users] How to check how many G729 
codeclicenseinstalled 
g723 gsmulawalawg726 
adpcmslin lpc10g729 
speexilbc g723 
- - 
- - 
- - 
- - 
- - 
-gsm 
- - 
3 3 
4 3 
2 9 
- - 131 
ulaw - 
5 - 
1 3 
2 1 
8 - - 
130 alaw 
- 5 
1 - 
3 2 
1 8 
- - 130 
g726 - 
6 3 
3 - 
3 2 
9 - - 
131adpcm 
- 5 
2 2 
3 - 
1 8 
- - 130 
slin - 
4 1 
1 2 
1 - 
7 - - 
129lpc10 
- 8 
5 5 
6 5 
4 - 
- - 133 
g729 - 
- - 
- - 
- - 
- - 
- -speex 
- - 
- - 
- - 
- - 
- - - 
ilbc - 
9 6 
6 7 
6 
512 
- - 
- this means i have no g729 codec 
installed.. thanks guys! 
:p On 11/13/05, Gentian Bajraktari  [EMAIL PROTECTED] 
wrote: Do: *CLI show 
translations If you see - (lines) on the G729 
row/columns than you do not have any G729 support. 
 RG. - 
Original Message - From: "Sahil Gupta" [EMAIL PROTECTED] 
To: "Asterisk Users Mailing List - Non-Commercial Discussion"  
asterisk-users@lists.digium.com 
Sent: Sunday, November 13, 2005 1:03 PM Subject: Re: 
[Asterisk-Users] How to check how many G729 codec  
licenseinstalled  Right 
:)   Regards,  
  Sahil Gupta  VoiceValley 
   On Sun, 13 Nov 2005, Angelito Manansala 
wrote:   *CLI show g729 
 No such command 'show g729' (type 'help' for help) 
  this means i have no g729 codec installed, 
right?On 11/13/05, Zafer Khodr 
[EMAIL PROTECTED] 
wrote:  That's easy...  Just 
go into asterisk cli and type" show g729" 
  It will tell you how many are active and how many 
you have in total  
  Regards  
Zafer-Original 
Message-  From: [EMAIL PROTECTED] 
 [mailto: 
[EMAIL PROTECTED]] On Behalf Of 
 Angelito  Manansala 
 Sent: Sunday, 13 November 2005 10:31 PM 
 To: asterisk-users@lists.digium.com 
 Subject: [Asterisk-Users] How to check how many G729 codec 
license  installed  
  Guys, is the any CLI commands or info files where 
you can check how  many g729 codec 
 license installed.  
   Regards,  
Lito  
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--  Best Regards,  Angelito 
Manansala  www.voicefidelity.net 
 Mobile: +639175425807   DID: (+63) 44 
7906770  msn: [EMAIL PROTECTED] 
 skype: bulcrack  
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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-13 Thread Gentian Bajraktari 

Do:
*CLI show translations

If you see - (lines) on the G729 row/columns than you do not have any G729
support.


RG.

- Original Message - 
From: Sahil Gupta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Sunday, November 13, 2005 1:03 PM
Subject: Re: [Asterisk-Users] How to check how many G729 codec 
licenseinstalled




Right :)

Regards,


Sahil Gupta
VoiceValley

On Sun, 13 Nov 2005, Angelito Manansala wrote:


*CLI show g729
No such command 'show g729' (type 'help' for help)

this means i have no g729 codec installed, right?

On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:

That's easy...
Just go into asterisk cli and type   show g729  
It will tell you how many are active and how many you have in total


Regards
Zafer

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelito
Manansala
Sent: Sunday, 13 November 2005 10:31 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to check how many G729 codec license 
installed


Guys, is the any CLI commands or info files where you can check how
many g729 codec
license installed.


Regards,
Lito
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--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-13 Thread Gentian Bajraktari 

Yes.

- Original Message - 
From: Angelito Manansala [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Sunday, November 13, 2005 1:23 PM
Subject: Re: [Asterisk-Users] How to check how many G729 codec 
licenseinstalled




g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
  g723 - - - - - - - - - - -
   gsm - - 3 3 4 3 2 9 - -   131
  ulaw - 5 - 1 3 2 1 8 - -   130
  alaw - 5 1 - 3 2 1 8 - -   130
  g726 - 6 3 3 - 3 2 9 - -   131
 adpcm - 5 2 2 3 - 1 8 - -   130
  slin - 4 1 1 2 1 - 7 - -   129
 lpc10 - 8 5 5 6 5 4 - - -   133
  g729 - - - - - - - - - - -
 speex - - - - - - - - - - -
  ilbc - 9 6 6 7 6 512 - - -


this means i have no g729 codec installed..

thanks guys!

:p


On 11/13/05, Gentian Bajraktari [EMAIL PROTECTED] wrote:

Do:
*CLI show translations

If you see - (lines) on the G729 row/columns than you do not have any 
G729

support.


RG.

- Original Message -
From: Sahil Gupta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 1:03 PM
Subject: Re: [Asterisk-Users] How to check how many G729 codec
licenseinstalled


 Right :)

 Regards,


 Sahil Gupta
 VoiceValley

 On Sun, 13 Nov 2005, Angelito Manansala wrote:

 *CLI show g729
 No such command 'show g729' (type 'help' for help)

 this means i have no g729 codec installed, right?

 On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:
 That's easy...
 Just go into asterisk cli and type   show g729  
 It will tell you how many are active and how many you have in total


 Regards
 Zafer

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Angelito

 Manansala
 Sent: Sunday, 13 November 2005 10:31 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] How to check how many G729 codec license
 installed

 Guys, is the any CLI commands or info files where you can check how
 many g729 codec
 license installed.


 Regards,
 Lito
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 --
 Best Regards,
 Angelito Manansala
 www.voicefidelity.net
 Mobile: +639175425807
 DID: (+63) 44 7906770
 msn: [EMAIL PROTECTED]
 skype: bulcrack
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--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Re: [Asterisk-Users] Last CVS - High Load

2005-07-12 Thread Gentian Bajraktari 



Yes I have experienced the same on my test machine. 
It has been like this for 3 weeks of CVS Head.
Someone must havea look at that, I think is 
the SIP channel.



  - Original Message - 
  From: 
  Thierry Wehr 
  To: asterisk-users@lists.digium.com 
  
  Sent: Tuesday, July 12, 2005 10:37 
  AM
  Subject: [Asterisk-Users] Last CVS - 
  High Load
  
  Good 
  morning
  
  on our Test 
  Machine based on RedHat 9 on a Pentium 4 SMP we are experiencing a 99% CPU 
  load for asterisk with the CVS of tonight
  
  does anyone 
  noticed that
  
  best 
  regards
  Thierry
  
  

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Re: [Asterisk-Users] Problem with asterisk+gnugk

2005-05-31 Thread Gentian Bajraktari 
You can either download the executable version of gnugk or you can reinstall 
the other versions of the pwlib and openh323 as they are only needed during 
the compile.


RG,

Gentian

- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, May 31, 2005 8:14 AM
Subject: [Asterisk-Users] Problem with asterisk+gnugk





Hi!
I'm trying to build gnugk with asterisk. Asterisk is working well with 
chan_h323

built with needed PWlib v.1.5.2 and open H.323 v.1.12.2.
But gnugk' s installing instructions says that I need latest PWlib(1.17.1) 
and

openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's
compiling fails and I get error 1.

Do you have any working solutions with asterisk and gnugk and what are 
needed

version numbers which you use to get then work together?

Thanks in advance!




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Re: [Asterisk-Users] Ztdummy usage

2005-05-31 Thread Gentian Bajraktari 



You can build it alone.
Then try to 'modprobe zaptel' and then 'modprobe 
ztdummy'
If you do this without errors it must 
work.
For more info read the wiki.

RG,
Gentian


  - Original Message - 
  From: 
  Mohamed A. Gombolaty 
  To: asterisk-users@lists.digium.com 
  
  Sent: Tuesday, May 31, 2005 12:22 
PM
  Subject: [Asterisk-Users] Ztdummy 
  usage
  Dear All, 
  I have installed Asterisk everything is OK until I tried to configure 
  meeting room, configuration was simple enough when I try I get a message that 
  it's not a valid meeting room, Now I don't have a Zaptel device on my 
  machine, so I found that you will have to use ztdummy to make a dummy 
  zaptel device on your machine and this is because of timing issues. 
  My question is ztdummy can only be done when making asterisk or is ther a 
  way to do it after post installation? 
  I am using by the way freebsd 5.3, built Asterisk from the ports 
  successfully. --
Thx
MAG 
  
  

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Re: [Asterisk-Users] sip extension logon failed problem

2005-05-26 Thread Gentian Bajraktari 

Which IP are you using on the client configurations for proxy server?

Gentian

- Original Message - 
From: lanfei chen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, May 26, 2005 4:08 PM
Subject: Re: [Asterisk-Users] sip extension logon failed problem



Hi Gentian,
 Thank you so much.
 But it doesn't work.
 My problem is in the case of bindaddr=0.0.0.0
If I set bindaddr=LAN-IP, it cannot logon to its sip
proxy server.
If I set bindaddr=WAN-IP, the sip extensions cannot
logon.

 Did you test and succeed? Wish you can give me more
information.

Thank you.
Be well.

lanfei


--- Gentian Bajraktari [EMAIL PROTECTED]
wrote:

Check if the SIP.conf is configured to bind into the
ip address of  LAN or
not? You have to define the internal address there,
not the WAN ip address.

RG,

Gentian

- Original Message - 
From: lanfei chen [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 25, 2005 2:52 PM
Subject: [Asterisk-Users] sip extension logon failed
problem


 Hi All,
Now I configured a linux box as a router. And I
 installed Asterisk on it.
My problem is whenever the WAN is offline all
the
 sip extensions will logon failed. My sip
extensions
 are connected to Asterisk through LAN. Why the LAN
 side sip phones cannot logon when WAN is offline.

 Who knows the reason and solution?
 Thank you so much.
 I really appreciate.

 Be Well.

 lanfei



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Re: [Asterisk-Users] sip extension logon failed problem

2005-05-26 Thread Gentian Bajraktari 
Well you must only use the LAN IP as the proxy ip when configuring the 
clients. Then use the LANIP to be the bindaddress at sip.conf. This will 
make the SIP proxy available to the intermal clients even if the WAN 
interface goes down.


Gentian
- Original Message - 
From: lanfei chen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, May 26, 2005 5:55 PM
Subject: Re: [Asterisk-Users] sip extension logon failed problem



several IPs,
67.105.168.51,
67.105.168.52,
61.235.77.130,
218.104.232.147

Thank you for your response.

lanfei

--- Gentian Bajraktari [EMAIL PROTECTED]
wrote:

Which IP are you using on the client configurations
for proxy server?

Gentian

- Original Message - 
From: lanfei chen [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 26, 2005 4:08 PM
Subject: Re: [Asterisk-Users] sip extension logon
failed problem


 Hi Gentian,
  Thank you so much.
  But it doesn't work.
  My problem is in the case of bindaddr=0.0.0.0
 If I set bindaddr=LAN-IP, it cannot logon to its
sip
 proxy server.
 If I set bindaddr=WAN-IP, the sip extensions
cannot
 logon.

  Did you test and succeed? Wish you can give me
more
 information.

 Thank you.
 Be well.

 lanfei


 --- Gentian Bajraktari [EMAIL PROTECTED]
 wrote:
 Check if the SIP.conf is configured to bind into
the
 ip address of  LAN or
 not? You have to define the internal address
there,
 not the WAN ip address.

 RG,

 Gentian

 - Original Message - 
 From: lanfei chen [EMAIL PROTECTED]

 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 25, 2005 2:52 PM
 Subject: [Asterisk-Users] sip extension logon
failed
 problem


  Hi All,
 Now I configured a linux box as a router.
And I
  installed Asterisk on it.
 My problem is whenever the WAN is offline
all
 the
  sip extensions will logon failed. My sip
 extensions
  are connected to Asterisk through LAN. Why the
LAN
  side sip phones cannot logon when WAN is
offline.
 
  Who knows the reason and solution?
  Thank you so much.
  I really appreciate.
 
  Be Well.
 
  lanfei
 
 
 
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site
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Re: [Asterisk-Users] sip extension logon failed problem

2005-05-25 Thread Gentian Bajraktari 
Check if the SIP.conf is configured to bind into the ip address of  LAN or 
not? You have to define the internal address there, not the WAN ip address.


RG,

Gentian

- Original Message - 
From: lanfei chen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, May 25, 2005 2:52 PM
Subject: [Asterisk-Users] sip extension logon failed problem



Hi All,
   Now I configured a linux box as a router. And I
installed Asterisk on it.
   My problem is whenever the WAN is offline all the
sip extensions will logon failed. My sip extensions
are connected to Asterisk through LAN. Why the LAN
side sip phones cannot logon when WAN is offline.

Who knows the reason and solution?
Thank you so much.
I really appreciate.

Be Well.

lanfei



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Re: [Asterisk-Users] Asterisk real time extensions problem...

2005-05-19 Thread Gentian Bajraktari 



HI,

The problem is that you are using: 
incoming-next,60069,1
Use: incoming-next|60069|1 
instead

RG,
Gentian



  - Original Message - 
  From: 
  Bharat M. Sarvan 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Cc: asterisk-dev@lists.digium.com 
  
  Sent: Sunday, April 17, 2005 11:52 
  AM
  Subject: [Asterisk-Users] Asterisk real 
  time extensions problem...
  
  
  Hello 
  everybody,
   
  I have setup asterisk real time extensions and its working pretty well. But 
  the problem is when I am jumping between the contexts using the Goto statement 
  in the database. I am getting a error 
  
  = Parsing 
  '/etc/asterisk/sip_notify.conf': Found
   -- 
  SIP Seeding peers from Astdb: 'ezzibpo4' at [EMAIL PROTECTED]:5061 
  for 60
   -- 
  Executing Goto("SIP/ezzibpo4-4636", 
  "incoming-next,6069,1")
  May 19 05:00:04 
  NOTICE[6420]: pbx.c:1688 pbx_extension_helper: Cannot find extension '6069' in 
  context 'incom'
  May 19 05:00:04 
  WARNING[6420]: pbx.c:6256 ast_parseable_goto: Priority 
  'incoming-next,
  
  The structure of the extensions db 
  is as given below
  
  ++---+---+--+-+--+
  | id | 
  context | exten | 
  priority | 
  app | 
  appdata 
  |
  ++---+---+--+-+--+
  | 1 | 
  incoming | 6069 
  | 1 | 
  Goto | 
  incoming-next,6069,1 |
  | 2 | 
  incoming | 6069 
  | 2 | 
  Hangup 
  | 
  |
  | 3 | incoming-next | 
  6069 | 1 | 
  DigitTimeout | 
  10 
  |
  | 4 | incoming-next | 
  6069 | 2 | ResponseTimeout | 
  30 
  |
  | 5 | incoming-next | 
  6069 | 3 | 
  Background | 
  welcome 
  |
  
  
   The 
  context “incom” in the above error is the context defined for placing outgoing 
  call in the sip.conf file. I don’t understand why is it looking for extension 
  6069 in the “incom” context.
  
   The 
  “Goto” statement in the context incoming is getting executed without any 
  probs, but the control is not getting transferred to the context 
  “incoming-next” upon execution of the Goto statement. 
  
   
  
   
  Could anybody suggest me as to where might the problem be and any way to get 
  rid of this problem. Please do reply….
  
  
  
  
  Regards,
  Bharat 
  M. Sarvan
  
  
  

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