Displaying 20 results from an estimated 400 matches similar to: "BT Caller ID - From Patch ?"
2004 Sep 27
1
G729 Private Licensing ??
Is anyone selling G729 License elsewhere other than Digium?
Anyone allowed to sell a similar License as a reseller?
-Kannaiyan
2004 Aug 13
11
asterisk in india
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com)
2004 Jun 14
7
collaboration with Panasonic PBX
Hi.
I've searched the archives and found nothing regarding collaborating
Asterisk with a Panasonic PBX (TD1232 to be exact)
Here's my question:
Can I use a Wildcard X100P to connect an outgoing line jack (on the
Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk,
and calls from Asterisk to the PBX?
On the hardware page for the X100P card is says it's great for
2004 Sep 13
2
allowing/disallowing codecs in dialplan?
Hi all,
Is there a possibility to set the codecs Asterisk will choose in the dialplan
("exten=>" statements or their contexts) instead of sip.conf?
My problem is that I connect my SIP phone with several providers (Nikotel,
Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers
offer the same set of codecs. I'd like Asterisk to use the same codec for the
2004 Jan 23
3
UK BT Interface with asterisk?
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
2003 Dec 18
6
G729 question
I am thinking about using the G729 codecs on my endpoint devices and
purchasing some G729 licenses for Asterisk but I have several questions:
1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I?
2. If I have G729A on one end and G729B on the other, are they compatible?
I have looked all over the place for question 2, but without buying the
ITU docs
I cannot seem to find this
2003 Dec 18
2
Zaprtc compile error - virtual device for conferencing
Hi,
I don't have a zaptel device for conferencing.
I read from the lists, that
ztdummy and zaprtc need to be installed to get conferencing.
I could able to compile successfully with ztdummy and when i receive the
call it says,
-- Goto (13732,s,1)
-- Executing MeetMe("SIP/-08118800", "1234") in new stack
== Parsing
2004 Jan 18
2
Asterisk as SIP Redirect Server -- Implemented - Not Working - Plz Help
I have coded chan_sip.c so that you can have
// sip.conf
register => username:password@somedomain.com/redirectconfig
[redirectconfig]
redirect=yes
redirecturi=sip:12345@domain1.com
redirecturi=sip:34556@domain2.com
redirecturi=sip:87877@domain3.com ....
so when you receive a call it will redirect to the alternating uri's with a
SIP 300 Message.
It works with the following sequence,
2003 Dec 16
2
Unable to Receive Fax -- RxFAX Application
Hi,
Below if the error message which I got from asterisk.
I was trying to fax to asterisk from my fax machine. I really dunno what
is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone
help me what could be the problem.
-- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack
-- Goto (13732,s,1)
-- Executing
2004 Jun 24
1
Delay in Zap Calls?
I have this line in my extensions.conf,
exten => _393.,1,Dial(ZAP/${EXTEN:3},20,tr)
when I make a zap call, it gives me two rings and then makes the zap call.
Is there is a way I can make the call immediate?
Kannaiyan
2004 Jan 14
5
SNOM IAX image
Hello.
I've been going through the archives, but can't discern the state or future direction of IAX on the SNOM100.
The most recent image appears to be from September 2002.
There was a message on this list stating that SNOM was coming to visit Digium last April with the intention of adding IAX support themselves.
For a while there was reference to the I100E on the asterisk and/or
2004 Sep 18
3
uk caller id
dear all, i am looking to enable CALLERID on an Asterisk system comprising a
X101P FXO interface connecting to BT PSTN in the uk
seems this is supported by the interface but there seems to be varying
information on how to enable it in zapata.conf
1. usecallerid=uk
2. ukcallerid=yes
being two of the configuration statements offered
TIA
GT
2004 May 22
14
Caller ID with BT CD50
Hi All,
Having searched the archives, I can see there has been much discussion
at various points regarding capture of caller id information from good
old BT.
If I understand correctly, it seems that not only do the drivers not
currently support it, but my X101P possibly/probably can't do it anyway
due to hardware?
So, that leaves me with the modem route, which seems more and more
unlikely,
2004 Jul 11
20
New Asterisk bounty: SIP simultaneous
>When I call a SIP user, the phone should ring in more
than one
>extentions. Also more than one phone should be able to
register with
>asterisk. Right now it is not the case.
There is no issue here. You seem to be confused, that's
all.
A SIP account is a SIP account and an extension is an
extension. You can assign an extension to an account (or
to multiple accounts) and the tool for
2004 Apr 30
6
app_dbodbc segfault
Is anyone out there using app_dbodbc
(http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it?
I was able to get it all working, but it causes * to segfault every now
and then. It does not appear to be related to any specific function
(ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I
noload the module, everything works fine, but when its running, after
calls to any of the
2004 Sep 28
7
UK (British Telecom) Caller ID again
I've followed the recent thread on caller id with UK British Telecom
networks (where the caller id data is delivered before the first ring).
My understanding is that if I use a recent CVS head (e.g.
CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to
do is include the line:
usecallerid=uk
In my zapata.conf (in the [channels] section)
I've done this, but I get:
Sep
2005 Aug 30
1
X100P and UK CallerID
Hi,
I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the
current gentoo ~x86 versions), with the UK CallerID patches from
http://www.lusyn.com/asterisk/patches.html applied.
The Zap interface itself seems to work fairly well - although it's a
little quiet, there is no echo. Unfortunately, there's also no
CallerID.
My zapata.conf is as follows:
[channels]
2006 Jun 23
1
SIP -> PSTN calls not connecting properly
Hi,
I've got a problem with my asterisk set up which has been going on for a
while (months). I'm currently running 1.2.7.1 on a gentoo box with the
topology below:
+-----+
PSTN ---------+ * +------------- Service Provider
(wctdm400p) +-+-+-+ IAX
| |
| |
FXS --+ +-- SIP (cisco 7940)
2004 Jan 23
1
Back to front logging for calls placed through /var/spool/asterisk/outgoing?
I've just noticed that if you start a call by writing a file to
/var/spool/asterisk/outgoing the cdr created on termination logs the call
placed to the local extension - not to the destination in the PSTN. Hence
there is no record of the PSTN number dialled. I guess most people want to
log the outgoing portion not the local call leg? Anyone know of a setting
that changes this?
Iain
2004 Jan 23
2
Latest cvs * compile error anyone?
I downloaded asterisk and was trying to compile fresh, It end up in
error, Any help appreciated.
cvs checkout asterisk
cd asterisk
make clean
make
END UP with following error, (Previously I was able to compile without
any errors. After a make clean stopped compiling.)
gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o
-lmysqlclient -lz
/usr/bin/ld: cannot find -lmysqlclient