Hello,
Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method
'bri_cpe when module load chan_dahdi.so.
Googling with chan_dahdi bri_net don't help much.
Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ?
Regards
___
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Hi Alejandro,
thanks for reply, interesting and I'll try it. $300 isn't that much if it's
reliable.
Dubravko
From: Alejandro Kauffmann [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote:
Hello,
Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method
'bri_cpe when module load chan_dahdi.so.
Googling with chan_dahdi bri_net don't help much.
Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ?
2008/11/27 Tzafrir Cohen [EMAIL PROTECTED]
On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote:
Hello,
Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling
method
'bri_cpe when module load chan_dahdi.so.
Googling with chan_dahdi bri_net don't help much.
Shall I
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
Regards
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I have an odd problem. I have just installed asterisk on an ubuntu
box, and migrated the previous configuration of asterisk (on another
ubuntu box) to this new server (scp -pr [EMAIL PROTECTED]:/etc/asterisk/*
/etc/asterisk/)
Asterisk worked fine on the old server, but on this server my SIP
trunk
I tried but no success. Do I have to add more to this?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
output is:
# strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony'
DAHDI Telephony
DAHDI Telephony Driver
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On Thu, Nov 27, 2008 at 1:03 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
This is not from 1.6, but rather from callweaver attached to Asterisk 1.4.
When i'll
On Thu, Nov 27, 2008 at 12:58:53PM +0100, Olivier wrote:
output is:
# strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony'
DAHDI Telephony
DAHDI Telephony Driver
A snippet from channels/chan_dahdi.c:
static const char tdesc[] = DAHDI Telephony Driver
#ifdef
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
1. Sip Config
Mode: Proxy
Primary Proxy IP Address: *.131
Primary Proxy port: 5060
Line1 Number: 1002
2. Security Config
Line1 Account:
On Thu, Nov 27, 2008 at 04:27:50PM +0500, Irfan Malik wrote:
I tried but no success. Do I have to add more to this?
What did you do? What did happen when you did that?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL
Hi All,
I want to prevent transfer on based of user,
means we can disable any user or peer to transfer calls in asterisk.
Can any one helps how can we prevent transfer feature.
I am using asterisk 1.4 branch.
Thanks,
Max Alex
Voip Developer
___
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On Thu, Nov 27, 2008 at 11:04:50AM +0100, Olivier wrote:
2008/11/27 Tzafrir Cohen [EMAIL PROTECTED]
On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote:
Hello,
Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling
method
'bri_cpe when module load
Ronald Wiplinger (Lists) wrote:
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
Hi,
as far as I can see, welltech also sells (embedded) asterix pbx units.
Chances are, you might find a manual
2008/11/27 Tzafrir Cohen [EMAIL PROTECTED]
On Thu, Nov 27, 2008 at 12:58:53PM +0100, Olivier wrote:
output is:
# strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI
Telephony'
DAHDI Telephony
DAHDI Telephony Driver
A snippet from channels/chan_dahdi.c:
static const
Thanks for this detailed reply.
I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with
either iaxmodem or t38modem.
Have you tried any of those 2 (iaxmodem or t38modem) ?
Which one would you pick ?
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El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió:
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
I've one wellgate 3804 (old version) with 4 fxo ports integrated with
Olivier wrote:
Thanks for this detailed reply.
I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax
with either iaxmodem or t38modem.
Have you tried any of those 2 (iaxmodem or t38modem) ?
Which one would you pick ?
iaxmodem only does audio FAXing (for the present).
Just to follow-up, because this may one day be found by someone with the
same issue, I fixed this:
My problem was that my sip peers did not have a call-limit setup. For some
(unknown to me) reason, hints only work for peers with a call-limit defined
(if using realtime, that would mean
Hi there!
Trying to originate and dial a number using Zap-8, used to work, but now it
just fails.
I enabled all debug I found in the source-code and this is the output from
asterisk.
Can someone understand something from the debug-output what is wrong and direct
me to what the problem might
On Thu, Nov 27, 2008 at 05:02:17PM +0100, Johan Sandgren wrote:
Hi there!
Trying to originate and dial a number using Zap-8, used to work, but now it
just fails.
I enabled all debug I found in the source-code and this is the output from
asterisk.
Can someone understand something from
On Thu, 28 Dec 2006 12:34:46 -0600, Savoy, Kevin - Williston, ND wrote:
checking for mysql_init in -lmysqlclient... no
What do I need to make that say yes?
You need to read config.log and check _why_ the link fails.
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Mike, I don't want to be a smart ass, but (as you claimed) if you didn't
change anything
I've had Asterisk and Polycom phones work perfectly with hints for
the last 6 months. Suddently, I realize they've stopped working in the
last few days. I haven't changed the configuration in any way.
You can either add that feature to chan_iax2.c or pay someone to add
that feature to chan_iax2.c.
Bruno Castelo Branco wrote:
Somebody know some work around for it?
I still trying to find a solution but nothing seems to work
thanks
Eric ManxPower Wieling wrote:
The problem is that IAX2
On Thu, Nov 27, 2008 at 4:39 PM, Olivier [EMAIL PROTECTED] wrote:
Thanks for this detailed reply.
I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with
either iaxmodem or t38modem.
Have you tried any of those 2 (iaxmodem or t38modem) ?
Which one would you pick ?
We
Is working on 1.6.0.1?? someone was able to make it work?
Thanks!
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Hi guys,
I have 1 zap channel in my house shared among couple people. If someone dials
911, I want that zap channel to be disconnected right away to make way for the
911 call.
I dug through voip-info.org and didn't find much.
Any hints?
kel
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I'm trying to get my Windows Mobile 6 phone working as an asterisk client.
Overall things are working well. However, I regularly get the following
message:
[Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote
host can't match request NOTIFY to call
'[EMAIL PROTECTED]'.
I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I
did the register test since I am using the realtime DB and asterisk does
periodic quesries to it for each registered user. Although Asterisk
continued to function as usuall, it was in a steady loop querying the DB for
the
Hi Sebastian,
http://bugs.digium.com/view.php?id=11196
Nguyễn Đình Trung
---
QiS Technologies, ltd.
Tel: 0168 528 7522
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Guillermo Salas M. wrote:
El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió:
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
I've one wellgate 3804 (old version)
Have you snniffed the packages? It seems to be some kind of difrerence
on the notify, try to sniff a packet ok and then one with error
Enviado desde mi iPhone
El 28/11/2008, a las 01:01 a.m., OCG Technical Support
[EMAIL PROTECTED] escribió:
I’m trying to get my Windows Mobile 6 phone
Thanks for the answer i did everything that is on that issue but is
not working, do you have it working? Every thing else im doing real
time but moh never check the db if i try the command realtime load ...
I get the values just fine, but they are never realy load to the
memory classes
Hi All
I have one issue regarding override callerid when i have anonymous call.
I have added PAI in sip header and also set sendrpid = yes in sip.conf
but the callerid is not overriding while i am sending call to three digit
calling like 911.
please give some idea and help for this issue!
I am
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