Hi
I use asterisk with Realtime/Mysql.
I have put a Call-limit at 1, but if the SIP account receive two call in
same time,
the second call don't ring busy.
Ayone know a solution for the second call get a busy ring ?
Best Regards
Jerome
--
Hi
a new small question ;=)
We have two Asterisk, connected in IAX2.
On the first, in dialplan, we have:
exten = _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
we sent into the IAXVAR ACCOUNTID the accountcode.
On the second, in dialplan, we have:
exten =
Hello,
I am looking to know if it is possible to modify the SQL query that is
on Realtime sip accounts.
I want multiple servers use the same sql table, so getting an extra
server field to indicate that the data is valid on the X server
is this possible?
thank you in advance
jerome
--
Hi
A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2
and now all SIP Relatime user are rejeted :
[Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming:
Received REGISTER (2) - Command in SIP REGISTER
[Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658
:
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with MySQL
Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with MySQL.
That's work and in my extension.conf, i have:
[as5300-incoming]
switch = Realtime
and in extconfig.conf
extensions = mysql,general,VOIP_Extensions
A lot
Thanks, it's limited the number of table ?
Le 01/10/2010 11:07, Захаров Антон a écrit :
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am
Le 01/10/2010 11:10, Steve Howes a écrit :
On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
That's work and in my extension.conf, i have:
[as5300-incoming]
switch = Realtime
and in extconfig.conf
extensions = mysql,general,VOIP_Extensions
A lot
After test, that's don't work :=
Le 01/10/2010 11:07, Захаров Антон a écrit :
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a expert
Hi
anyone know what is a the solution of this problems ? :
[Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to
[Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to
anyone know this error message ?
Phibee Network Operation Center a écrit :
Hi
anyone know what is a the solution of this problems ? :
[Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to
[Nov 20 13:04:45] WARNING[12043
qualify=yes
trunk=no
notransfer=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
Phibee Network Operation Center a écrit :
anyone know this error message ?
Phibee Network Operation Center a écrit :
Hi
anyone know what is a the solution of this problems ? :
[Nov 20 10:15:40
to authenticate as 04NUMBERCALLED
Phibee Network Operation Center a écrit :
No change, now:
voip*CLI iax2 show peers
Name/UsernameHost Mask Port
Status
Trader-Classic/ 78.SERVER2 (D) 255.255.255.255 4569 (E) OK
(2 ms)
1 iax2 peers [1
Hi
My first post get no answer :=, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
Hi
i have a small problems on two Asterisk Server 1.6.4 :
The first sent the call to the second, and in the second, i have a error :
[Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call
rejected, CallToken Support required. If unexpected, resolve by placing
address
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
but into my extensions.conf:
exten =
Hi
I have finished the installation of my VoIP basic configuration ...
Actually:
- All calls from my E1 are received by a Cisco AS5300 and sent to my
Asterisk (in G711 by SIP).
- All user are connected by SIP to the Asterisk
- All calls from User are sent by asterisk to the Cisco
Hi
I Use Asterisk 1.6.1 with Realtime and a mySQL database,
Actually, my extensions.conf are:
===
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp
Hi
ok i have understand ;=)
bye
Phibee Network Operation Center a écrit :
Hi
actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
I read on the wiki:
===
Database Config
put the following in res_mysql.conf
[general
Hi
when i use MeetMe, i have this errors:
app_meetme.c: Unable to open pseudo device
Where is the problems ?
i have too warning and error into my logs:
[Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open
pseudo channel for timing... Sound may be choppy.
[Nov 1 07:26:17]
Hi
actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
I read on the wiki:
===
Database Config
put the following in res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = myuser
dbpass = mypass
dbport = 3306
Phibee Network Operation Center a écrit :
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '042600' rejected because extension
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '042600' rejected because extension not found.
[Oct 28 06:01:18] NOTICE[12813]:
Hi
i test a new equipment on my backbone: a Cisco AS5300 with voice dsp
ressource
connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and
all Asterisk outgoing
call are sent to Cisco AS5300.
Actually, i configure the AS5300:
isdn switch-type
Hi
I search to know if a company or user use that a Cisco AS5300 with DSP/Voice
with Asterisk ?
I want use the AS5300 only for the E1/PSTN link in/out
thanks Jpc
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Hi
anyone know where i can find all internatinal numbering plan in csv and
for free or small price ?
thanks
Jpc
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Register Now:
Anyone use CIsco 1760 with Asterisk
Phibee Network Operation Center a écrit :
Hi
i am search a sample config (for asterisk and for cisco) for connect
a cisco 1760 with a FXO card to my asterisk.
Thanks for your help
Jerome
Hi
thanks for your answer,
not curious, i have one 1760 with FXO card and
~100 Cisco 1751 with FXO Card to at connected to my asterisk in SIP
but i don't have touch Asterisk since 18 mounth ... and never connected
router
at asterisk (only Linksys SPA941 voice unit)
if you have a idea of the
Hi
i am search a sample config (for asterisk and for cisco) for connect
a cisco 1760 with a FXO card to my asterisk.
Thanks for your help
Jerome
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asterisk-users mailing list
To
I have a problem connecting a Grandstream ipphone to an asterisk.
The ipphone is behind a nat router, I redirected UDP 5060 and 5004 to my
phone.
It connects well to the asterisk server. I can call outside and receive
calls from outside without any problems.
But if I call from this ipphone to
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