Displaying 20 results from an estimated 3000 matches similar to: "asterisk canreinvite option questions"
2007 Jun 25
2
Rining 180 and 183
Dear all
I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya
[asterisk]-----[mediant 2000]--------[Avaya]
when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of
2007 Sep 17
2
Filesharing + video + voice supported Soft phone
Dear all
I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any soft phone which support this all feature ?? with asterisk
Regards
Satish Patel
---------------------------------
Moody friends. Drama queens. Your life? Nope! - their life, your story.
Play Sims Stories at Yahoo! Games.
2007 Jul 30
6
outbound caller ID
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a voip provider (surely)
Thanks,
Vieri
____________________________________________________________________________________
Moody friends. Drama queens. Your
2007 Aug 23
1
speex payload value
hmm...forgive my ignorance here. icould have explained it wrong.
the rtp header has the pt (payload) field as a 7 bit value. i was under the
impression speex had a particular value i should set it to. is this so? if
no what value should i assign it, whether by convention or otherwise?
Note that i'm implementing a simple rtp header and combining it with the
speex payload i'm not using
2007 Jun 20
2
zlib1g
Hi List;
Why I need zlib1g to do installation for Zaptel? Will
zlib1g do compression or it will what extactly do
during the installation process?
Regards
Bilal
____________________________________________________________________________________
Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.
http://sims.yahoo.com/
2007 Aug 06
2
ATA phones ring when they register
Hi,
I have an 8-port Grandstream GXW-4008 V1.2A ATA
converter with analog phones connected to it.
They work fine except for just one "feature" I would
like to modify. Somehow, each time the ATA
re-registers the SIP clients or each time the device
has to be rebooted for maintenance, the phones ring
once. This feature can be useful as it notifies the
user of the re-registration.
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
Hi List;
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I know that if I place the host with static IP then no
need to register, but what if the voip gateway was
having dynamic IP and I do not need to register on
asterisk, but I
2007 Sep 24
0
Oracle CLusterware fails while runing root.sh
I am installing Oracle Clusterware 10.2.0.1 on Linux 4.5 ES on OCFS2 and ASM. my ocr and oss files are on ocfs2. I got following error while running it. Please let me know what went wrong.
--------------------------------------------------------------------------------------------------------------------
[root@linux2 crs]# ./root.sh
WARNING: directory '/u01/app/oracle/product' is not
2007 May 13
2
extracting text contained in brackets ("[ ... ]") from a character string?
I have a text string that contains text within two brackets.
e.g. "testdata[3]" "testdata[-4]", "testdata[-4g]",
I wish to "extract" the string enclosed in brackets?
What is a good way to do this?
e.g.
fun(testdata[3]) = '3'
fun(testdata[-4g]) = '-4g'
---------------------------------
Moody friends. Drama queens. Your life?
2007 Jul 23
6
phone directory with asterisk
Dear all
I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ?
Rgd
satish patel
2007 Oct 09
3
Monit reporting that pid changed
Hi, I was wondering if this is normal
or if it something i need to wonder about.
I get Monit alerts that my mongrels PID files
have changed.
Changed Service MONGREL_2
Date: Tue, 09 Oct 2007 14:19:38 +0000
Action: alert
Host: sa.greenling.com
Description: ''MONGREL_2'' process PID changed to 18368
Your faithful employee,
monit
2007 Sep 28
1
RPM package wish list
hello,
Does our beloved Centos project have a page where one can request that an RPM package be built, especially for C5? How about alerts when those requests have been satisfied?
I know about 3rd party repos like rpmforge, epel, and kbextras. I can search those in YUM.
Centos seems to have specialized as a server OS in the past, but my experience is that it is an up and coming
2007 Oct 03
1
Asterisk doesn't answer to incoming call
Hi:
I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension.
I'd apreciateany idea.
---------------------------------
Moody friends. Drama queens. Your life?
2007 Aug 11
2
speex and rtp
Hi all,
i've been having problems for the past little while with finding a app library
for my voip application. then looking at speexclient i noticed there are
timestamps and sequence numbers included in the implementation.
my question is: should i get a third party RTP lib for my application or
simply emulate the speex client in its RTP like properties. wouldn't it
be easier and
2007 Jul 13
2
standardization
Hi
I have dataframe which contain 5 columns and 1000 records. I want standard each cell.
I want range each column between 0 and 1 . I think i must use loop?
could you help me?
---------------------------------
Moody friends. Drama queens. Your life? Nope! - their life, your story.
[[alternative HTML version deleted]]
2007 Aug 20
2
Firefly IAX2 configuration
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the Asterisk, I did the following configuration on
the /etc/asterisk/iax.conf:
[iax2user1]
type=friend
context=internal
username=iax2user1
secret=password
2007 Jun 19
1
Error handling
Hello,
I have a question about error handling. I run simulation studies and often the program stops with an error, for example during maximum likelihood. I would like the program not to stop but to continue and I would like to ask how the error handling can be set up for this (if it can). I tried to look through manuals etc but unfortunately did not get closer to the solution. Below is a
2007 Aug 08
1
[fdo] error on cross compiling libdbus
List,
I want to cross compiling libdbus for arm processor
with Montavista toolchain. But I went into error.
Says,
checking for posix getpwnam_r... configure: error:
cannot run test program while cross compiling
Commmand I use is
configure --host=arm-linux
--prefix=/root/Desktop/bluez-3.13/dbus-glib-0.72
CC=/opt/montavista/mobilinux/devkit/arm/v6_vfp_le/bin/arm_v6_vfp_le-gcc
2007 Aug 23
3
[Bridge] bridge problem when one interface is in blocking mode
Hi,
We have a simple bridge setup but the ping (and other
network traffic) does not work reliably. After tracing
the code, it looks like a software bug. Since bridge
software is been running by thousands of people. I
guess I am wrong. Anyway, here is the problem.
There are 2 boxes and each one has 2 interfaces, 1
ethernet and 1 wifi. STP is enabled for the bridge to
avoid the loop. So the box 1
2007 Oct 05
0
asterisk-users Digest, Vol 39, Issue 12
Ok.. will be there...
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: Thursday, October 04, 2007 12:50 PM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 39, Issue 12
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