Displaying 20 results from an estimated 5000 matches similar to: "Still having hangup problems in NZ"
2005 Sep 12
2
Hang up not hanging up (New Zealand Indications??)
Hi there,
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
I have found this information:
http://bugs.digium.com/bug_view_page.php?bug_id=0001474
Which discusses my problem and i have made sure that i have the latest
info in the indications.conf as follows:
[general]
2005 Sep 20
6
iax2 trunking wackyness
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd, maybe there is a version
skew?
Any suggestions will be appreciated.
Thanks
Clive
2005 Jan 17
3
On Hold music
This may sound kind of crazy and I maybe missing something. But are you
placing the call on hold so you can hear the hold music. This may not
be the case but you may have to place the call on hold to here the
music.
Also you mentioned sound, you do not need a sound card in the asterisk
box to use this hold music feature.
Hope this helps.
-----Original Message-----
From:
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2005 May 25
4
SER Help
Hi,
I'm looking for a tutorial or installation guide for
SER to be used with asterisk to solve the remote SIP
agent problem. All the documents available are for
large scale installation.
Any help is highly appreciated.
Regards.
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2005 Mar 15
2
Wiki down: Is there another source for documentation?
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
Thanks!
Sean
2005 Sep 30
2
Why does the s extension not work in my extensions.conf file
Hello
In my extensions.conf file:
[frompstnisdn]
exten => s,1,Dial(SIP/200&SIP/202,20)
exten => s,2,Voicemail(su200)
exten => s,3,Hangup
I use the s, start, extension to handle incoming calls.
In my zapata.conf:
context=frompstnisdn
This works ok on another asterisk box I setup. But on incoming calls I get:
-- Extension '787367' in context 'frompstnisdn'
2005 May 16
4
Web Client with IAX2 and ilbc
Guys.
Maybe this is asking for a lot :) but is there any web client that can use
IAX2 and ilbc?
This is for a "call us" web idea.... Any leads?
2005 Sep 30
2
analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
Hello
We have setup a doorbell which has an inbuilt analog phone which is
connected to our Asterisk via a SPA2000 ATA. The problem we are getting is
that when a caller presses the buzzer it is taking two or more minutes to
finally call the reception phone.
In the SPA2000 I have set dtmfmode to be inband.
I notice that with the asterisk you dial a number and then it waits for a
timeout
2004 Sep 21
12
Astricon pictures
Hey,
I am here at Astricon and about to go down to registration. Is there any
interest in pictures if I take my digital camera? I am sure that someone
is already doing this. (Probably someone official). I would take
pictures of each day and upload them to my website if anyone is
interested. Let me know!
--
Kristian Kielhofner
2006 Jan 11
3
video development
Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2,
but I guess that's not the answer you are looking for.
If you manage to do this and release it under GPL I'll kick in $50 for a
bounty.
Regards,
Dean Collins
dean@collins.net.pr
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
2005 May 25
5
Asterisk Crashing; Not getting Core dumps
This is frustrating. Asterisk has crashed now twice today and neither crash
has produced a core file. My ulimit is unlimited.
I'm using safe_asterisk so asterisk is restarting immediatly, but how the
hell am I suposed to find out wtf happened with no core file? Debug log
doesn't say anything either.
AGRHHHHHHHH
-Matthew
--
2006 May 26
4
mpg123 or asterisk
should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
also, has anyone succedded in compiling mpg123 in a dual core pentium
with centos 4.3 ?
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
2005 Jan 13
3
High delay with diax099f + Asterisk
Hi all!
Somebody knows something to do with a high delay using Asterisk + DIAX!?
When I used IAXComm(Linux) in both sides(peer and me) no problems.
Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the
voice coming from the person that I called. I don't have delay in my voice
to the peer phone.
CODEC: u-law (I tried with all available codecs)
Thanks for your help!
2003 Sep 11
0
Hangup Detection and BUSYDETECT_MARTIN
Hello,
I've got the following configuration:
2 X101Ps
Asterisk built with BUSYDETECT_MARTIN
busydetect=yes
busycount=10
callprogress=yes
signalling = fxs_ks
With this setup, the best I can do is get voicemail with 17 to 19 seconds of
silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has
anyone had any success with this?
It seems that hangups are indeed detected,
2005 Aug 28
2
Need quote for Asterisk and billing remote install
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
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2005 May 23
4
Digium FXS modules too fragile?
Hi all,
Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.
2005 Aug 28
4
Mplayer as replacement to mgp123 in MP3Player cmd
There is a patch to mplayer that allows it to suppress stdout messages
and instead output pcm data to stdout. I managed to get it working with
app_mp3.c and seems like it is working fine. All that was needed was a
change in the execl line and a slight increase in timeout value. I have
only done limited testing. If mplayer is able to replace mpg123 without
issues, this opens up a whole lot of
2006 Apr 26
6
Sphinx2
I have a gateway, which I call from my mobile phone (free of charge,
since it is the same phone company).
This gateway gives me a dial tone. I can than dial to any extension
number or even other gateways, ....
It is getting more a trouble to remember all the numbers, or to key in
all the long phone numbers when you got the dialtone.
I was thinking of using for this Sphinx2. How can I
2005 Jun 05
2
TDM400P Polarity reversal detection
Hi
Can TDM400P detect polarity reversal on FXO module?
We have C.O. lines that reverse polarity on Answer and release.
Thank You
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