hello every body
i have problem in receiving fax from e1 lines. this is my scenario:
faxphoneericson pbx ---e1asterisksip-zoiper-softphone
when i send fax from zoiper, i can receive it successfully on the faxphone
but when i send fax from faxphone, i can not receive it on
hello everyone,
i have question about fax detection on dahdi channels. does dahdi channels
detect fax and pass it? if yes, does it detects both types of fax (g711
pass through and T.38)? finally, how can i enable it on dahdi_channels? i
set faxdetect=both in chan_dahdi.conf but dahdi can not pass
hello everybody
i want to send fax via asterisk in pass through mode. everything is ok if
enable fax detection in ooh323 and write fax extension in extensions.conf
file. just one problem: delay. i have to wait 5 seconds in order to fax
detection done. it is too long for me when i have voice call
:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *s m
*Sent:* Wednesday, May 20, 2015 2:54 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] reduce delay in fax detection
hello everybody
i want
hello every body,
i have big problem to configure h323 trunk between cisco router and
asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module
can work with cisco routers or not (in gateway mode, it is ok and
register in cisco gatekeeper but i can not configure trunk h323)
Ending Call Monitor thread
cisco debug shows rtp message with src address 0.0.0.0. i really don't know
how i should fix it. please help me.
thanks
SAM
On Wed, May 6, 2015 at 10:44 AM, Dmitry Melekhov d...@belkam.com wrote:
06.05.2015 10:06, s m пишет:
hello every body,
i have big problem
. is
there any difference between g711 codecs which cisco and asterisk utilize?
On Wed, May 6, 2015 at 11:56 AM, Dmitry Melekhov d...@belkam.com wrote:
06.05.2015 10:58, s m пишет:
Hello!
I'm not h323 expert, may be somebody else can understand from this log
what is happening, but I can't
hello every body,
i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
ooh323 module. i configured both side and have successful call from cisco
to asterisk. but when call comes from asterisk to cisco, my phone rings but
no audio is heard and call is disconnected after 5
hello everybody,
i want to configure a sip trunk between my system which has asterisk 11.5.1
and a cisco router. this is my scenario:
Freepbx-my system-cisco-routerFreepbx
my system acts like a router. if i set just one codec in dial-peers on
cisco router, every thing is ok and i
hello everybody,
i want to have authentication on RAS messages between gatekeeper and
gateway. i have a cisco gatekeeper and an asterisk gateway. is it possible
to have h235 on asterisk gateway in order to send authenticated RAS message
to gatekeeper? if yes, how can i add it to my asterisk? i am
hello all
i want to have overlap dialing in asterisk. it works fine if i don't have
! in my pattern. for example pattern 07. works fine and i can call
07122 by overlap dialing via it. but if i define 07! i can't call 07122
because it doesn't wait to collect all digits and therefore call 07. if
hello guys
i want to slimming my asterisk by loading only mandatory modules. in order
to do that, i edit my modules.conf file and set autoload=no and load just
mandatory modules.
my problem is, how should i determine which modules are necessary to
asterisk works correctly? i have sip, h323 and
hello every one
i want to have multiple sip calls with different codecs for each one. for
example call to 8100 has g729 codec while call to 7900 has ulaw codec.
i searched a lot and found that there is some variable like sip_codec
which can set codec for a special inbound or outbound call. i don't
hello all
i'm using ooh323.so module for my h323 connections and it works fine. i
just have problem with loading and unloading module. you know, ooh323
module doesn't support reload command. it means, if ooh323 module is loaded
and i reconfigure my h323 channels (add another channel), i should
the call
playing voice prompts etc.
On Sat, Oct 12, 2013 at 9:52 AM, s m sam.gh1...@gmail.com wrote:
thank you everybody for your useful replies and so sorry to answer late.
i understand what i need. first of all, i wanna to use pass through g729
codec (which is free). so i go to http
thank you everybody for your useful replies and so sorry to answer late.
i understand what i need. first of all, i wanna to use pass through g729
codec (which is free). so i go to http://asterisk.hosting.lv/ to get g729
codec. i have freebsd 8.2 and asterisk 1.8.22 but there is no compatible
thank you Dominik you help me a lot.
and the last question is how many license key should i buy? i read that
license for g729 is per-channel but i don't understand what channel exactly
means here. this is my scenario :
10endpointspbx181...pbx182...pbx183...10endpoints
pbx181 and pbx183 has
then you dont need any license. Just download passthrough g729
license.
Mitul
On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com
wrote:
On 10/02/2013 09:33 AM, s m wrote:
and the last question is how many license key should i buy? i read
that license for g729 is per-channel but i
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when i
run core show codecs in asterisk, there is a g729 codec in the list so i
assume that i can use it for my channels. but connection can not be set
when i use it for my h323 channel.
i read somewhere that codec g729
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
s m sam.gh1...@gmail.com schrieb:
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when
i
run core show codecs in asterisk, there is a g729 codec in the list
so i
assume that i can use it for my channels
hello all
i have asterisk 1.8.22 and have problem with caller id. this is my
scenario:
PSTN -- FXO --- FXS --- phone(223)
when i call from a 223 to another phone, every thing is ok and caller id
(223) is shown in called phone. but when i call from another phone to 223,
no caller id is shown and
hello all,
i have a conceptual question.
i have a h323 gateway and it is connected to a h323 gatekeeper. my
question is: can i connect my gateway to another gateway directly? i mean
can these two gateways work with each other without working with
gatekeeper? or when i have connection with a
hello all,
i want to have ooh323 connection between asterisk and cisco. in my
scenario, asterisk is gateway and cisco is gatekeeper.
this is my ooh323.conf file:
[general]
port=1720
bindaddr=192.168.0.227
gateway=yes
faststart=yes
h245tunneling=yes
h323id=g...@test.com
settracelevel=10
hello all,
i want to have ooh323 connection between asterisk and cisco. in my
scenario, asterisk is gateway and cisco is gatekeeper.
this is my ooh323.conf file:
[general]
port=1720
bindaddr=192.168.0.227
gateway=yes
faststart=yes
h245tunneling=yes
h323id=g...@test.com
settracelevel=10
hello every one,
i have an asterisk system and want to act as gateway and send calls to
cisco gatekeeper.
this is my h323.conf file:
[general]
port=1720
binaddr=192.168.0.YY
context=from-trunk
faststart=yes
h245tunneling=yes
gatekeeper=192.168.0.XX //cisco address
progress_setup=8
hello everyone
i have a simple question: i have an asterisk which is a h323 gateway
and has a h323 connection to a cisco gatekeeper and a sip connection
to a pbx.
my question is: how can i send all calls to gatekeeper?
i searched a lot and found that i should set gatekeeper=192.168.0.X
(ip
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
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
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
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
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
/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${EXTEN:1
suggestion.
thanks
sam
On 4/11/13, Asghar Mohammad asghar...@gmail.com wrote:
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${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
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
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
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