Displaying 20 results from an estimated 800 matches similar to: "3-way calling"
2004 Sep 24
2
Asterisk as PSTN gateway
I've been asked to recommend a solution for a one-E1-port PSTN gateway
supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know
they work. I use the Asterisk software in a couple of places and would like
to use the E100P. My question is whether anyone out there has any
installations using this and what their opinion is about it (does it work?
how's the audio quality?
2004 Nov 25
1
astcc newbie question
I'm trying out ASTCC. I set the card length to 10, and generated a test
card. 10 digits. I set the extensions file to:
exten => 9175954700,1,Answer
exten => 9175954700,2,DeadAGI(astcc.agi)
exten => 9175954700,3,Hangup
I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits.
How come it thinks it is 12 digits?
I set both the Published number and DID in the Brand
2004 Dec 14
3
Problems with app_realtime
It seems that when setting qualify = 200 or qualify = yes in the database for
a sip friend/peer, RealTime does not update the registration status like it
should.
I also have several peers which have been offline and Asterisk still reports
them as registered, even though the registration seconds are only 200.
Asterisk Ver: CVS HEAD 12/1/2004
Layout of sip_buddies:
mysql> describe
2004 Aug 11
5
Asterisk and SMP
Does anything have to be done at compile time in order for Asterisk to
take advantage of 2 CPU's?
Thanks
2005 Feb 18
2
Sending DTMF after a call is set up
I'm using Dial to place a call to a PBX. But then I want to wait a few
seconds and dial an extension. Dial doesn't return until the call is
disconnected though.
I also want the caller to not hear any audio until the DTMF has been sent.
This gets the caller to the right place and he doesnt have to hear the
welcome message from the PBX.
Dial apparently isnt the application to use. Is
2004 Dec 24
3
Preventing Asterisk from sending 'h' across to SIP Provider
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI scripts
on hungup
2004 Jun 30
7
Asterisk Causing Cisco 7200 Router to Crash?
Hi,
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always when Asterisk starts up. Has anyone
else seen this problem? It is very odd because this is a very good router and
we
2005 Jan 10
1
Ramifications of Multiple Sip Reloads Within Minutes?
We have the ability to create random UID's on own system through a custom
CGI API. These UID's are written to individual sip configuration files based
on the account name, so for instance sip_TEST.conf, sip_TEST2.conf, and
sip_TEST3.conf, etc. Many of these UID's are created on the fly and at random
times throughout the day. Right now, I have it setup to do a reload every
night.
2004 Dec 06
1
SIP status lagged
Hi,
When I do a sip show peers in the cli, the status is lagged.
This peer its behind a satellite link with 600/900ms of delay.
May I change some parameter in the Asterisk?
Some times I cant make a phone call from the remote site to my central site.
Thanks
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2005 Jan 17
1
ASTCC single stage + no access number + auth usingsip username and password
> I would like to have all SIP phones to work on prepaid basis
> and without need to dial any access number, instead I would
> like to use the phone as normal dialing only the destination
> number, for example 00464090510.
I use the AccountCode for authentication. This is how, for example:
exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2})
> Once the call is
2005 Jan 12
2
Trouble building appradius
I am having trouble building appradius from http://appradius.minitelecom.org/
I configure, make, make install cpprad-1.0, but when I configure, then
make appradius I get :-
obelix:/usr/src/appradius/appradius1.0 # make
make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory
2004 Jun 25
2
Problems Compiling and Loading asterisk-oh323 0.6.2
Hi,
I having a problem compiling the wrapper for oh323. I am running Debian,
kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the
openh323 version I have is 1.13.5. I execute the following commands first
before attempting to compile the wrapper:
pwlib_1.6.6:
make both
openh323 1.13.5
./configure
make opt
asterisk-oh323 0.6.2
make
2004 Sep 28
1
Sep 28 17:52:28 WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 2bda648025dbf8c52fd293515d98d2c2@216.252.176.45 for seqno 102 (Non-critical Request)
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2005 Jan 03
9
Just saw your [Asterisk] xJack Segfault in Asterisk
Hi:
Just saw your post while trying to solve a similar asterisk problem. Did
not see any responses. Was your problem solved and what was the solution?
Carey
2004 Dec 28
1
Sending e-mail from dialplan
I would like help with a "dial plan" that will do the following: I feel
pretty good about each of the elements except; how to e-mail the
recorded file to an e-mail address.
Allow a caller to call into the system:
1. Answer
2. play a short pre defined greeting
3. Allow caller to enter "PIN" during the Item #2 greeting
a. If the caller entered THE valid pin (1 system
2006 Feb 08
1
New trick for old dogs
We have been using Samba for many years. The company has just switched from an NT domain to an Active Directory domain. The new server is running Windows Server 2003. We are having trouble configuring our Solaris 8 server so it can join the domain as a server. Just getting Samba to compile and link was interesting enough. This included downloading and compiling a new version of the BerkeleyDB,
2004 Dec 14
3
Realtime problem
I'm having trouble with the Realtime setup. I've followed the instructions on
voip-info using odbc but I get this message during asterisk boot:
Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory)
Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load
config sip.conf, SIP disabled
== Registered channel type 'SIP' (Session
2006 Sep 29
2
hcc not found, rcmd build
Working under Windows XP, I am compiling a package called 'pgirmess'
with the command
rcmd build --binary --auto-zip pgirmess
I have this message error after having listed: functions text html latex
example chm
....
zipping help file
hcc: not found
cp: cannot stat 'c:/TEMP/Rbuild365620874/pgirmess/chm/pgirmess.chm': No
such file or directory
make[1]: *** [chm-pgirmess] Error 1
2008 Sep 19
1
readRegistry function (PR#12937)
Full_Name: Zivan Karaman
Version: 2.7.2
OS: Windows XP
Submission from: (NULL) (195.6.68.214)
I'm puzzled by the readRegistry function.
Shouldn't the "hive" argument be something like
c("HLM", "HCR", "HCU", "HU", "HCC", "HPD") rather than
c("HLM", "HCR", "HCU", "HU",
2017 Feb 07
2
Clang option to provide list of target-subarchs.
There are at least four clang frontends for offloading to accelerators:
1 Cuda clang 2 OpenMP 3 HCC and 4 OpenCL. These frontends will
want to embed object code for multiple offload targets into a single
application binary to provide portability across different subarchitectures
(e.g. sm_35, sm_50) and across different architectures (e.g nvptx64,amdgcn).
Problem: Different frontends