Hi
On Fri, Nov 10, 2006 at 10:45:27PM +0100, Christian wrote:
Hello,
OK, this is what i have done. This is a newly installed Debian the latest
testversion.
Everything is done as root.
Before I did this I upgraded the system to the latest by doing:
apt-get dist-upgrade
Rebooted.
apt-get
Hi,
exten = 6000,1,Dial(SIP/6000,15,tr)
exten = 6002,1,Dial(SIP/6002,15,tr)
exten = 6004,1,Dial(SIP/6004,15,tr)
exten = 6006,1,Dial(SIP/6006,15,tr)
exten = 6008,1,Chanspy(SIP/6006 |wbq)
when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one.
when coversation
Todd- Asterisk ha scritto:
I'm preparing to deploy a small number of Grandstream BT101's and
GXP2000's to a remote location (which I won't have access to). I'd
like to have them pull a config file from my server - I'm almost there...
The phones are looking for the config file on my webserver
hi,
i got following message
please tell me what can i do after this ..
linux:~/asterisk-gui # make checkconfig
--- Checking Asterisk configuration to see if it will support the GUI ---
* Checking for http.conf: OK
* Checking for manager.conf: OK
* Checking if HTTP is enabled: OK
*
On Sat, Nov 11, 2006 at 05:22:49PM +0530, Thirumal Saminathan wrote:
hi,
i got following message
please tell me what can i do after this ..
Post a question in [EMAIL PROTECTED] ?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
Gustavo,
Glad to help.
Gustavo, Linux and Asterisk are tools for implementing a telephony
system, so you must to know telephony basics first. Fortunately Asterisk
will force you to learn telephony!!.
Regarding transfers, see the following scenario. A calls B and B
transfer the call to C.
Blind
Andrew,
Could you explain what problems do you have with Hitachi 5000?. We have
carried out extensive tests with Hitachi 5000 at customer location who
is planning to install more than 120 wifi phones. It is a mining
company at 4200 mts altitude, covering the mining camp, an small village
and
http://www.voip-info.org/wiki-Asterisk+config+features.confOn 11/11/06, Ronald Wiplinger
[EMAIL PROTECTED] wrote:I want to add some sound filed on demand during a phone call only
possible on some extension numbers.I get many phone calls from local companies, but don't understandChinese! I would
Hello,
I have the following call file:
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27
MaxRetries: 2
RetryTime: 10
Context: test2
Extension: s
And the following dialplan:
[test1]
exten = s,1,NoOp(${CALLERIDNUM})
But my CALLERIDNUM and CALLERIDNAME variables are both empty. I tried
without
Hi,
No, I still get that error as before. And I havent installed anything special.
Many thanks,
Christian
On 2006-11-10 at 23:08 brandon kruz wrote:
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
cd zaptel
make clean ; make distclean ; sh configure ; make ; make install
modprobe
Christian,
Could you be out of disk space? What is the output of df -k and mount.
Also does /root/zaptel-1.4.0-beta2/tonezone.h exist? Assuming the
source is at that directory level.
Bob...
Christian wrote:
Hi,
No, I still get that error as before. And I havent installed anything
Hello,
Using 1.2.13 with bristuff:
exten = 8599,1,Answer()
exten = 8599,n,Wait(1)
exten = 8599,n,MusicOnHold(default)
Whan the call comes through a zap (telco) channel I can't hear the
music, but through a sip/iax channels I hear it.
Any idea why?
Thanks,
Arik Raffael Funke wrote:
Hello,
I have the following call file:
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27
Caller id format is:
CallerID: SomeName SomeNumber For example, I use:
CallerID: VM-System 4200
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
On Sat, Nov 11, 2006 at 05:33:24PM +0100, Christian wrote:
Hi,
No, I still get that error as before. And I havent installed anything special.
Many thanks,
Christian
Right. So I guess that there's no point repeating on that the 10-th
time. How about trying to answer some more focused
Hi Anthony -
Has anyone noticed that attempting to place a call from the Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes
simply returns the phone to the idle screen? It is not related to the
number being dialed, as we have observed two entries for the same
number,
On Fri, 2006-11-10 at 17:29 -0800, Tom Lynn wrote:
Add me to the list. Not only lagged, but also failures to register.
AND, apparantly Paypal won't automatically authorize payments to them
anymore. I'm not recharging my account anymore.
Is working fine from me. You can reach the payments
On 2006-11-10 15:48:23 -0800, Andrew Joakimsen [EMAIL PROTECTED] said:
I am surprised that you have had good success perhaps you haven't done
proper testing?
I see you are skeptical...
I am using the Nokia e60, which also has no problems on the asterisk
side. The phone could use some
Hi,
See my answers below.
On 2006-11-11 at 19:15 Tzafrir Cohen wrote:
I repeat: please give the output of:
ls -la tonezone.h /usr/include/zaptel
/usr/include/zaptel:
total 72
drwxr-xr-x 2 root root 4096 2006-11-11 17:18 .
drwxr-xr-x 101 root root 8192 2006-11-11 04:30 ..
lrwxrwxrwx 1
I thought google was my friend ? Does this mean I have two friends now ?
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 10, 2006 8:44 PM
Subject: Re: [asterisk-users]
Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Take a look at http://www.jivesoftware.org/ - perhaps some way you can use that?
rajeevOn 11/10/06, Andrea Spadaccini [EMAIL PROTECTED] wrote:
Ciao Ondrej, That's why I was more thinking about mysql
I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this .
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On Sat, Nov 11, 2006 at 08:07:35PM +0100, Christian wrote:
Hi,
See my answers below.
On 2006-11-11 at 19:15 Tzafrir Cohen wrote:
I repeat: please give the output of:
ls -la tonezone.h /usr/include/zaptel
/usr/include/zaptel:
total 72
drwxr-xr-x 2 root root 4096 2006-11-11
On Sun, Nov 12, 2006 at 12:51:27AM +0530, Rajeev Natarajan wrote:
Or you can look at PHP-AGI; use the php to query mysql (probably more
scalable than dialplan MYSQL)
Running an external php script which will open a separate mysql
connection, query it, close and be done is not exactly scalable.
Christian,
Either mkdir -p /usr/include/zaptel/tonezone or delete the tonezone.h
link then re-run the build. The date stamp on the link does not
correspond with the others the directory. Could be something leftover
from an earlier attempt?
Bob...
Christian wrote:
Hi,
See my answers below.
Hi
i have to forward a call from my asterisk server on another server but
my server is behind nat.
How can i setup my extension.conf?
Actually i have set up it as follows:
exten = 046566,1,Dial(SIP/[EMAIL PROTECTED])
my server has a private ip 192.168.100.249 and doesn't have a public ip
Hmmm accesisng the GUI using the browser?
Seriously: did you encounter any problems?
l.
On Sat, 11 Nov 2006 12:52:49 +0100, Thirumal Saminathan
[EMAIL PROTECTED] wrote:
hi,
i got following message
please tell me what can i do after this ..
linux:~/asterisk-gui # make checkconfig
On 2006-11-10 09:10:30 -0800, Mario François Jauvin
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried
with or without ztdummy device,
Hi,
Finally, its working!
Many thanks for all your help.
If I want to get the latest Asterisk through SVN, should i use the trunk
version as well?
many thanks,
Christian
On 2006-11-11 at 21:42 Tzafrir Cohen wrote:
On Sat, Nov 11, 2006 at 08:07:35PM +0100, Christian wrote:
Hi,
See my answers
On Sat, Nov 11, 2006 at 10:50:06PM +0100, Christian wrote:
Hi,
Finally, its working!
Many thanks for all your help.
If I want to get the latest Asterisk through SVN, should i use the trunk
version as well?
many thanks,
Christian
At this point the latest is branches/1.4 , unless you're
On 21:44, Sat 11 Nov 06, Tzafrir Cohen wrote:
On Sun, Nov 12, 2006 at 12:51:27AM +0530, Rajeev Natarajan wrote:
Or you can look at PHP-AGI; use the php to query mysql (probably more
scalable than dialplan MYSQL)
Running an external php script which will open a separate mysql
connection,
Regardless, they're still perpetually lagged. I'm suspicious as to why paypal is conducting a review. For now, considering the poor performance, I stand by my decision to shop the market.
On 11/11/06, Vicky [EMAIL PROTECTED] wrote:
I doubt how many days more voxee will survive . Its been a month
Then I guess I'd better hurry up and use my remaining 49 cents worth of credit!!On 11/11/06, Vicky [EMAIL PROTECTED]
wrote:I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this .
Martin Joseph escribió:
On 2006-11-10 09:10:30 -0800, Mario François Jauvin
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried
with or
Hi,
I havent found any sounds in the svn version of Asterisk 1.4. If I download the
tarball form their FTP there are sounds in it. Any thoughts?
Many thanks,
Christian
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asterisk-users mailing
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
bye
Ronald Wiplinger
On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I want to add some sound filed on demand
On Sat, Nov 11, 2006 at 09:42:27PM +0200, Tzafrir Cohen wrote:
I wonder, though, how that symlink was created. I hope an advice by me
was not involved...
I guess that more than just advice. zaptel-source and libtonezoe-dev
versions 1.2.9.1.dfsg-2, 1.2.10.dfsg-1 and 1.2.10.dfsg-2 contain the
The sounds and music files are added through the menuselect process. Do make menuselect after the .comfigure and select the sond files. They are then pulled on the ftp site.On 11/11/06,
Christian [EMAIL PROTECTED] wrote:
Hi,I havent found any sounds in the svn version of Asterisk 1.4. If I
I am running the most recent asterisk 1.2.13 on a Fedora 3.0.
When I go into asterisk (asterisk -r), defaults to verbose 3 and I get
a stream of messages:
Remote Unix connection
Remote Unix connection disconnected
...
...
(keeps on repeating).
I went to google and searched on asterisk Remote Unix
On 11/12/06, Yu Safin [EMAIL PROTECTED] wrote:
I am running the most recent asterisk 1.2.13 on a Fedora 3.0.
When I go into asterisk (asterisk -r), defaults to verbose 3 and I get
a stream of messages:
Remote Unix connection
Remote Unix connection disconnected
...
...
(keeps on repeating).
I
I've noticed sla.conf in Asterisk 1.4. I'd love to test it, but how does it work? There's bupkiss docs, and until I have a clue how to use it, I can't test it.
Did you ever find out anything on this? All I hear is people wanting us to test and test. How the heck do we test when we have no idea
Ron,The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording.
The best part of it is that you can hang up and the recording will
u need another box say box a with real/addressable ip address. create an iax entry in box a and have the private ip (box b) box register to box a. then you can do a Dial(IAX2/boxb/${EXTEN}) that will ring the extension connected to
your 192.168.100.249 boxhope that helps;) On 11/12/06, nik600
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