Displaying 20 results from an estimated 10000 matches similar to: "Call Connection Problem"
2006 Dec 04
2
ASterisk and SER
HI,
My Asterisk is registed with my SER. My client are connected to asterisk
when they dial any no like 62222 asterisk passes this is ser and then again
ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing
this exten it says loop detected. can some one tell me what is wrong.
thanks
arun
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2007 Jun 04
2
G729 License
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in my old box also ?
thanks
arun
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2007 May 13
2
TC400B load problem
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=0000000c, dsts=00000101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with
92 transcoders (srcs=00000101, dsts=0000000c)
May 13 14:56:36 pbx2
2007 Apr 24
6
Digium card sale
Good morning,
Pardon for this intrusion I just wanted to let everyone know about some of
the specials that I have going on at HYPERLINK
"http://www.astawerks.com"www.astawerks.com . From now until the end of
June I will have a huge unpublished sale on all Digium products. Prices are
way to low to list so I will have to be personally contacted. I also have a
permanent sale on all
2007 Apr 19
1
Asterisk Queue Call Transfer
Hi
I've configured the queue on my asterisk box and everything is working fine.
In my queue I've 3 agents logged in the queue. When call comes they are able
to receive the calls without any problem. But some time they are on break
and there extension rings and no one is there to answer the call (we don't
want them to log off from the queue) but we have one normal user in the same
2007 Apr 08
1
Adding Noise or background noise
Hi,
In my dial plan I've configured two trunks to make outbound calls (trunk1
and trunk2) to same service provider but I want when any of my exten starts
with _2. should goto trunk2 and there should be some kind of disturbance
(like some noise or some background noise) when my calls goes to trunk2 to
make the call quality bad. Mainly I want to achieve bad call quality on
trunk2 by adding
2007 Apr 17
2
No of Calls
Hi
sorry for asking the same question again:
here is my details:
I've 50 exten in my sip and I'm using snom300 to my asterisk box this
asterisk box is connected to another asterisk box using IAX trunk over 1MB
full duplex line. I'm using g729 as the preffered codec. Can you please tell
me how many calls can go at the same time without causing the any type of
problem.
thanks
arun
2007 Apr 20
1
CallerID Auth
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.
thanks
arun
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2009 May 20
3
Asterisk CCM, CME Integration
Hi All,
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME
-----> 461X Phones
461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X
Phones
so in
2007 Jun 06
2
iax trunking on OpenBSD
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing source?
kind regards
Sebastian
2007 Jun 27
1
Help with IAX Trunk
Hi
I've two servers :
1. UK
2. Pakistan
Pakistan * server has ISDN30.
Pakistan(ISDN30) <====> UK ===> User
Im planning to setup an IAX2 trunk between these two server ?
so , how much bandwidth I need for 30 simul. calls ?
Im planning to use G729 on both my server ?
to support 30 calls over IAX2 do I've to change some setting during compile
time or not ?
pls suggest.
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2007 Jul 17
5
Asterisk PRI Busy Problem
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my
2007 Apr 19
1
Ser as IVR
Hi,
Is it possible to design an IVR using SER ? If yes please advice.
thanks
arun
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2007 Apr 19
5
Polycom IP 501 is displaying wrong time
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
---------------------------------
Ahhh...imagining that irresistible "new car" smell?
Check outnew cars at Yahoo! Autos.
2007 Apr 22
3
Asterisk & M$ SQL Server
Hi all,
Has anyone successfully set up asterisk to query a M$ SQL Server?
I'd like to be able to query one in the dial plan and use the results to
tamper with call priorities / CLID etc.
If someone could point me to a howto / guide or relate their experiences
with this, that would be great !
Thanking you all in advance.
Callum
P.S. Before anyone asks; it's a legacy database that
2007 Apr 25
3
FYI
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net
Euro Tech News Blog http://eurotechnews.blogspot.com
2009 Oct 16
5
IVR
Ladies and Gentlemen,
We already have an Asterisk Call center suite installed at our contact
center. Now we wish to commence IVR services. We are offering Health
Information Services. Can someone help us to develop this Addon / Solution?
Best regards.
--
?????? ????? ????? ???? ???????
Nazir Ahmed Vaid
Cell:+92300-8277778
eHealth Services (Pvt) Ltd.
http://www.ehealth-services.com
NexSource
2007 Mar 07
4
OT Vonage V-Phone Adapter (Possible Hack)
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone. Looking at the pic, it
looks like the dongle is both a soundcard and memory stick. Heck, I
would be glad to have it if I could get the soundcard to work.
Might as well since it is free after rebate.
http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
2007 Apr 24
5
tone generation
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
If not, can I use some system command to generate the wav file
then just have asterisk play it?
Jerry