Displaying 20 results from an estimated 4000 matches similar to: "Timeout in Call Transfering"
2003 Jul 11
7
ISDN PRI E1 configuration with E100P
<P>hi Everyone,</P>
<P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P>
<P>*CLI> <BR> == D-Channel on span 1 up<BR> -- B-channel 1 successfully restarted on span 1<BR> --
2003 Jul 05
3
Activate MySQL logging
<P>hi,</P>
<P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P>
<P>Thank you inadvance,</P>
<P>Surajee</P>
<P> </P><br>
--------------This mail sent
2003 Aug 07
1
Warning Messages
hi,
i have connected a SNOM 200 to the asterisk. here are my settings,
Codecs
-------
Default codec - g.711u/g.711a
Packet size - 20ms
Negotiation - Interoperable
Type - 160
DTMF
----
Inband - negotiate
Outband - negotiate
Payload Type - 101
when a call comes to the SNOM or when making an outdial, following warning
messages are coming on asteisk,
WARNING[1209214400]: File dsp.c, Line 1198
2003 Jul 06
9
Accurate Billing
<P>hi everyone,</P>
<P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome
2003 Jun 11
3
Dialing out through a Hardware PBX
<DIV><FONT face=Arial size=2>hello All,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV>
<DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our
2003 Jul 18
16
Call Transfer
hi,
Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call.
ie, the operator answers the call, and presses hash key to transfer, and then enters the extension
number, some times, it timeouts too quickly before the operator enters the whole extension number
(may be bcos the operator is slow).
I tried the following, but it doesn't seems to be helping
2003 Sep 07
2
Call Time out Problem-Very Urgent!
hi,
I have a problem in call time out,
An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a
Nortel PBX is conneted to my server.
But when i do a Dialout(from both E1s)the calls do not timeout.
For ex.
Dial(Zap/g4/123456|20|t)
suppose other side is ringing and is not answering.
even after 20 seconds, call doesn't get timeout
pls gv me a solutions..
its really urgent..
Surajee
2003 Oct 16
1
Prob with Ringing multiple Channels
hi,
The prob is when we ring 2 channels simultaneously, only 1 channel is actually ringing.
In our configuration, the Asterisk box is connected to an E1 channel bank,
where 15 analog extensions are conencted to channelbank inturn.
We tried following,
Dial,Zap/g4/444&Zap/g4/448|20|t
Heres the output,
-- Executing Dial("IAX2[trunk10@trunk50]/1",
2004 Jul 23
0
qudBRI and transfering calls with the latest RC2.
I'm trying the latest bri 0.1.0 RC2 drivers.
In announce I see implementation of so long waited Transfer feature.
But I can't make it work.
When the person who is making transfer after talking with second party press
"R" second time to establish 3 way call
the person to which call supposed to be transfered being disconnected.
Any ideas whats wrong?
Thanks,
Dmitry
2003 Aug 07
2
Problem -ATA-711-723-Oh323-Asterisk
Hi List,
I am facing the reverse problem as stated here.I am
using ATA 186 to make
and recieve call to * through OH323 driver.
When I use G711 codec in the ATA to make call then
then as soon as i dial an
extension the * crashes with 'segmentation fault'.
But the same scenerio works fine when i use 723 codec
in the ATA .I can dial
the number and extension very well/(I have 723 support
in
2005 Aug 19
2
FXO not picking up; baffled
I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone
systems. Right now, I'm most certainly confused.
I have a TDM-04B (four FXO) and four analog FXO lines running into it
from an AdTran 616. I have Asterisk working internally, although I could
use some help getting incoming calls to answer properly and configuring
my outbound dialplan.
Here's where I'm
2004 Jan 27
1
Distinctive ring Issues
Hello all!
We have a PSTN line with four numbers calling into it. There is
distinctive ring on these lines. They are are follows:
1. standard ring
2. short ring
3. long ring
4. short ring, long ring, short ring
Based on the information I have been able to find, I have created the
following entries in my zapata.conf file, to
try and weed out some of the timings:
dring1=95,0,0
2003 Jun 01
6
Call Transfer Problem
hi All,
We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk.
We were able to do one type of call transfering, ie, the called person can transfer the original call to another person.
but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to
2005 Feb 11
1
Problem with # Transfer from queue
Hi I'm having trouble # transfering queue calls.
in extensions.conf I have:
[macro-queue]
;
; Places caller in queue
; ${ARG1} - Queue name to place caller in.
; ${ARG2} - Voicemail Extention
; ${ARG3} - Caller ID to Set.
exten => s,1,DBget(temp=nm/on) ; Get Night key, if not existing,goto 102
exten => s,2,Playback(custom/500/10)
exten => s,3,SetCallerID(${ARG3})
exten =>
2004 Aug 20
1
x100p won't answer
Hi,
I just got two digium x100p clones and installed asterisk on fedora
core 2 which took some tweaking. After getting asterisk up I installed
the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards),
which worked fine. ztcfg is showing two channels configured, but when I
start asterisk and do show channels, i see no active channels.
zapata.conf has:
signalling = fxs_ks
2003 Jun 24
0
Which type of lines to get from the Analog PBX??
<P>hi,</P>
<P>I have the following scenario,</P>
<P>an Asterisk server<BR>anolog pbx</P>
<P>I hav a channel bank, it accepts 30 analog lines coming from the analog pbx,<BR>the other end of the channel bank is the asterisk server with E100P card.<BR>I am confused as to which type of lines to get from the analog pbx.</P>
<P>I
2004 Jun 23
3
help needed with read()
Hi,
Greatly appreciate if some one help me with the application read().
asterisk*CLI> show application read
asterisk*CLI>
-= Info about application 'Read' =-
[Synopsis]:
Read a variable
[Description]:
Read(variable[|filename]): Reads a '#' terminated string of digits from
the user, optionally playing a given filename first. Returns -1 on hangup
or
error and 0
2004 Aug 21
3
zaptel config
Hi,
Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora
core 2 which took some tweaking. After getting asterisk up I installed
the zaptel stuff - then modprobed zaptel, wcfxo,
which worked fine. ztcfg is showing two channels configured, but when I
start asterisk and
2009 Feb 24
2
Multiple SIPGate accounts.
Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?
Thanks in advance,
Ray.
Excerpts from sip.conf -
[general]
8<---- SNIP! ---->8
Register => 1212121:aaaaaaaa at
2005 Feb 27
0
Interface * with ATA from ATA FXS port? (Here I go again)
Well, I thought I had my problem solved, but it is acting up again.
Hopefully this time I can provide enough information.
What I have is an * box setup with one X100P and TDM400 with one FXO and
one FXS. For my regular setup with interfacing with my PSTN and my
entire house with analog phones, the box is working great.
I am trying to interface a Mediatrix 1202 device to my * box via the