Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
thanks guys, i solve my problem.
as Asghar said, i remove 2 and forget to add it again therefore
asterisk can not recognize extension 200 in extension.conf file.
this is my extension that works properly:
exten=_2.,1,Dial(SIP/to-231/1${EXTEN:2})
thanks every body for your attention.
Sam
On 4/13/13, Gertjan Baarda gertjan.baa...@gmail.com wrote:
Can you post both extensions.conf from both systems?
Sent from my iPhone
On 11 apr. 2013, at 14:51, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf
?
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AJS
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
thanks Asghar, but it doesn't help. i have below error yet:(((
Dropping call because extensions '200', 's' and 'i' doesn't exists in
context [from-trunk]
i think that something is wring with my extensions in extensions.conf
but i don't know how to fix it.
please let me know if you have any other suggestion.
thanks
sam
On 4/11/13, Asghar Mohammad asghar...@gmail.com wrote:
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your
extensions.conf ?
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
why you are removing 2 and adding 2 ?
exten= _2.,1,Dial(SIP/to-232/here your are adding --2${EXTEN: here
you are removing 1st digit (2) -- 1})
try this exten= _X.,1,Dial(SIP/to-232/${EXTEN})
show me also sip users of both side.
let me know if this solve your problem.
On Sat, Apr 13, 2013 at 10:29 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar, but it doesn't help. i have below error yet:(((
Dropping call because extensions '200', 's' and 'i' doesn't exists in
context [from-trunk]
i think that something is wring with my extensions in extensions.conf
but i don't know how to fix it.
please let me know if you have any other suggestion.
thanks
sam
On 4/11/13, Asghar Mohammad asghar...@gmail.com wrote:
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your
extensions.conf ?
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
Can you post both extensions.conf from both systems?
Sent from my iPhone
On 11 apr. 2013, at 14:51, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf ?
--
AJS
Answers come *after* questions.
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[asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
hello all i,m newbie in asterisk and now want to sip and h323 connection. this is my scenario: phone(ext100)---freepbx---sip---system1---H323---system2---freepbx---phone(ext200) when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says service unavailable. i debug asterisk in my system 2 and see below message: Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk] i googled about this message and found that file extensions_mor_h323.conf should be included into /etc/asterisk/extensions_mor.conf. but i don't have any extensions_mor.conf file at all!!! is extensions_mor.conf really necessary to fix my problem?if yes, how i have connection in one way without this file? if no, how i can fix this problem? thanks in advance sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
On Thursday 11 April 2013, s m wrote: when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says service unavailable. i debug asterisk in my system 2 and see below message: Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk] OK. What do you have in the [from-trunk] context in your extensions.conf ? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf ?
--
AJS
Answers come *after* questions.
--
_
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your
extensions.conf ?
--
AJS
Answers come *after* questions.
--
_
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asterisk-users mailing list
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