date:20130424

Re: [asterisk-users] h323-sip: one way connection

2013-04-24 Thread s m

thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end


On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:

 try type=peer instead of friend.


 On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:

 i know what is the exactly problem. i enable debug for h323 and it says:
 could not find user by name 200 or address 192.168.0.146

 when i change peer-146 to 200 every thing is ok and i can call from
 two side. but it is not good for me because 200 is the name of extension
 and when i config asterisk systems, i don't know the name of extensions,
 therefore i should use addresses not name of extensions.
 do you know how i should define address of the other end in h323.conf
 file? i define the address by host=192.168.0.146 but asterisk can not
 find it? why?


 On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:

 please post cli output for both calls.


 On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:

 hello everybody

 i want to have sip connection between two asterisk systems (145 and
 146). connection from 145 to 146 is ok but i can not call from 146 to
 145.
 this is h323.conf file in 145:
 [peer146]
 host=192.168.0.146
 type=friend
 context=from-trunk


 [to-146]
 type=peer
 host=192.168.0.146
 faststart=yes
 tunneling=no
 progress_audio=yes
 disallow=all
 allow=alaw
 allow=ulaw

 this is mu extensions.conf file in 145:

 [from-trunk]
 exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
 [line-231]
 exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})

 i have this error: dropping call because extensions '100', 's' and 'i'
 doesn't exists in context default.

 if i change peer146 to general, every thing is ok and i can call
 from two side. my question is: in h323 connection, is it a MUST to
 have general context in h323.conf? if not, why i have this error and
 how i can solve it?
 thanks in advance
 sam

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Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

2013-04-24 Thread Steve Davies

On 4 April 2013 09:05, Ishfaq Malik i...@pack-net.co.uk wrote:

 On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote:
  On 03/26/2013 05:22 AM, Ishfaq Malik wrote:
   Hi
  
   In asterisk 1.8.7.0, an inbound call that was transferred to another
   peer would have 2 cdr entries.
  
   In asterisk 1.8.18.0 this same activity has a single cdr entry.
  
   This is a rather large and fundamental change to be enacting halfway
   through a single family branch, was there any reason why this happened?
   It means we can't upgrade without doing significant extra development
   and testing.
  
 
  This was most likely an unintended consequence of some other change
  (most likely dealing with masquerades). Is 1.8.18.0 the exact version
  when the behaviour changed?
 
  Just so I'm clear on the scenario, what are the channel technologies
  involved? Is the transfer initiated via a protocol message or via a DTMF
  feature?
 
  Thanks,
 
  Matt
 

 Hi Matt

 Did you ever spot/recreate the change I was referring to?



me too - I can confirm a behaviour change and will go try and pin down at
what point it happened.

This may take me a while :(

Steve
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Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

2013-04-24 Thread Ishfaq Malik

On Wed, 2013-04-24 at 12:48 +0100, Steve Davies wrote:
 On 4 April 2013 09:05, Ishfaq Malik i...@pack-net.co.uk wrote:
 On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote:
 
  On 03/26/2013 05:22 AM, Ishfaq Malik wrote:
   Hi
  
   In asterisk 1.8.7.0, an inbound call that was transferred
 to another
   peer would have 2 cdr entries.
  
   In asterisk 1.8.18.0 this same activity has a single cdr
 entry.
  
   This is a rather large and fundamental change to be
 enacting halfway
   through a single family branch, was there any reason why
 this happened?
   It means we can't upgrade without doing significant extra
 development
   and testing.
  
 
  This was most likely an unintended consequence of some other
 change
  (most likely dealing with masquerades). Is 1.8.18.0 the
 exact version
  when the behaviour changed?
 
  Just so I'm clear on the scenario, what are the channel
 technologies
  involved? Is the transfer initiated via a protocol message
 or via a DTMF
  feature?
 
  Thanks,
 
  Matt
 
 
 
 Hi Matt
 
 Did you ever spot/recreate the change I was referring to?
 
 
 
  
 me too - I can confirm a behaviour change and will go try and pin
 down at what point it happened.
 
 
 This may take me a while :(
 
 
 Steve
 --
Hi Steve

I raised a bug on this issue

https://issues.asterisk.org/jira/browse/ASTERISK-21394

I need to update it with the requested details though.

Regards

Ish

-- 
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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Re: [asterisk-users] h323-sip: one way connection

2013-04-24 Thread Asghar Mohammad

what flavor of h323 you are using?


On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:

 thanks Asghar,
 i do it, but no thing happened:(
 asterisk do not identify host line as ip address of the other end


 On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:

 try type=peer instead of friend.


 On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:

 i know what is the exactly problem. i enable debug for h323 and it says:
 could not find user by name 200 or address 192.168.0.146

 when i change peer-146 to 200 every thing is ok and i can call from
 two side. but it is not good for me because 200 is the name of extension
 and when i config asterisk systems, i don't know the name of extensions,
 therefore i should use addresses not name of extensions.
 do you know how i should define address of the other end in h323.conf
 file? i define the address by host=192.168.0.146 but asterisk can not
 find it? why?


 On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:

 please post cli output for both calls.


 On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:

 hello everybody

 i want to have sip connection between two asterisk systems (145 and
 146). connection from 145 to 146 is ok but i can not call from 146 to
 145.
 this is h323.conf file in 145:
 [peer146]
 host=192.168.0.146
 type=friend
 context=from-trunk


 [to-146]
 type=peer
 host=192.168.0.146
 faststart=yes
 tunneling=no
 progress_audio=yes
 disallow=all
 allow=alaw
 allow=ulaw

 this is mu extensions.conf file in 145:

 [from-trunk]
 exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
 [line-231]
 exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})

 i have this error: dropping call because extensions '100', 's' and 'i'
 doesn't exists in context default.

 if i change peer146 to general, every thing is ok and i can call
 from two side. my question is: in h323 connection, is it a MUST to
 have general context in h323.conf? if not, why i have this error and
 how i can solve it?
 thanks in advance
 sam

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Re: [asterisk-users] CDR Question

2013-04-24 Thread Nyamul Hassan

Hi,

On Sun, Apr 21, 2013 at 9:57 PM, jg webaccou...@jgoettgens.de wrote:

lastdata is the argument string of lastapp. Since Hangup() does not
have any args, lastdata is empty. If you need to store things that are
not already stored, you can either use the userfield or add extra columns
to the cdr table, which depends to some degree on how you store the call
data. Details can be found here: http://www.asteriskdocs.org/**
en/3rd_Edition/asterisk-book-**html-chunk/database_storing-**cdr.htmlhttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/database_storing-cdr.html

Thank you JG for your advice. I already understand why I am seeing a
blank. I was hoping a simple catch all type setting, that I can parse
from the CDR file by the script, rather than have to play around in the
Dialplan to get some variables in the CDR.

Regards
HASSAN

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Re: [asterisk-users] h323-sip: one way connection

Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

Re: [asterisk-users] h323-sip: one way connection

Re: [asterisk-users] CDR Question

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