Displaying 20 results from an estimated 2000 matches similar to: "911 settings."
2003 Dec 22
2
Sipura 2000 configuration.
Ok here is another problem I have run into.
I have a Sipura 2000 and I have been able to configure line 1 with only
one small problem. But I can't get the line 2 working with asterisk.
Here are samples of my sip.conf and extensions.conf. If I disable line
1 I can then get line 2 working. Is there a sample configuration for
the Sipura to get both ports working with Asterisk.
Sip.conf
2003 Dec 05
2
Help with setup IpDialog Sip Phones.
I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface.
Thanks
2003 Dec 03
1
Cisco and Asterisk 2621
Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right!
CISCO router model: 2621
VoIP module: NM-HDA-4FXS
I have done Google lookup and at the Wiki about
2003 Sep 05
0
Windows 2000 call viewer!
I am new to this forum. As well as a new user of Asterisk. My vendor installed the system and we are still trying to get all the bugs out of it! I have a few questions about configuration and a program to view who is on what extensions.
I am looking for a program that will work on my Receptionist work station. She is running Windows 2000 pro. We have not plans on upgrading to XP pro so
2003 Dec 19
1
Sip registration change!
I have a question on SIP devices that are setup and working but you
change the login name and contents to them why does asterisk need to be
shut down and restarted for them to work? I have reloaded extensions
and done a reload command. But the updated sip phones do not work until
I shut down and restart asterisk. Is there any other way to update them
without restarting the system? Since the
2004 May 18
0
snom 200 phones.
I have about 5 snom 200 phones working fine with everything. Voicemail,
Transfers and all. Except I can't seem to use them to pickup parked calls
nor place a call on park. I also have sipura-2000 with analog phones that
are able to pickup parked calls and to park them. Most of them are on
firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix
the problem. I get no error
2004 May 19
0
example of mulity company extension.conf needed.
I am trying to get a building that has 3 company's on one asterisk server.
I need to make the IVR via DID take them to there right menu. So far I have
everything working except when they goto via standard_marco to an extension
and are sent to voicemail they are dropped off in the first menu and not the
one they came from. In other word sent to another company's menu. If it
happens to be
2004 Jun 10
0
IAX Binding to 2 nic's for trunking two asterisk servers
I have a problem in that when you use IAX2 for trunking and have 2 nics one
is used to connect directly to 2nd Asterisk server how do we get the outside
Nic card to take IAX connections? Is there any way to get this working via
two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but
outside address section?
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2003 Oct 13
4
IAXTEL/ Dial problem
Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are.
CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack
-- Calling using options
2004 Jan 06
1
IAX2 Trunk two Asterisk boxes.
I need to get 2 Asterisk servers working together. I have been reading
and doing just about every example I have been able to find here on the
list and the Wiki. It's now gotten to the point that nothing on box2
seems to be working. I seem to have a major problem understanding the
format. Here is what I have so far. It's 3 days of hair pulling and
nothing seems to work!
Asterisk box 1
2003 Sep 17
4
Programming 976 numbers from dialing out.
I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help!
exten => 1900XXXXXXX,1,Congestion
exten => XXX976XXXX,1,Congestion
exten => XXX976XXXX,1,Congestion
exten => 1XXX976XXXX,1,Congestion
exten => 91900XXXXXXX,1,Congestion
exten =>
2003 Nov 24
4
Sip phones!
I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive!
1 - Sipura SPA-2000
2 - Grandstream Sip phone BT-102
1 - Grandstream HT-286
1 - Snom 105 Sip phone.
I have called and emailed chagres but they have not reply. Nor
2004 Jan 21
1
Sip phones transfer not working.
I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can not use the # key to do this due to we have several IVR's
2004 Jan 14
1
How do we updated to the new .7.1 version.
Yes folks it's me a Newbie. Remember I am also a non-Linux person trying
to learn. I have a production Server running Asterisk .5 12/02/03 CVS,
and would like to upgrade it to the new .71. Has anyone come up with
instructions (Documentation for us newbie) on how to do this? My
server is running Mandrake 9.0 which I know nothing about!
Sorry if this sounds stupid but all the instructions I
2003 Oct 07
3
Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says:
astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r
I have tried to release it with soft hangup Zap/1
& also soft hangup
2004 Jan 22
5
Snom 200 phones not working.
I have 2 Snom 200 and would like to get them to work properly with
Asterisk. With the Firmware 2.02t I am able to use the phone. But only
one line configured. With there newer firmware 2.03o it will not allow
me to make calls. But I can get calls on the unit. Again the 2nd line
is not able to be registered. Is this an issue with Asterisk or Snom?
I could use some example configuration
2004 Jul 22
1
Can anybody recommend a good T1/PRI provider ?
Are you looking for local or Long distance? What will these T1s primarily be
used for(inbound/outbound, domestic, local, long distance, international)
How important are per minute rates to you? how many minutes do you expect to
use per month?
We are in Tampa Florida and have 15 T1s from several different providers so
I may be able to refer you to one if it's a match to what you're
2003 Nov 12
3
DIAX 0.93 with some sound improvements and not only...
Hi all,
DIAX 0.9.3 is available for download from the same place:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
The new DLL contain the latest updates made by Steve in the iaxclient
library.
Still just IAX1 is supported (for the moment).
What's new in 0.9.3?
- accept blank passwords;
- accept for registration/calls host names, not only IP Address;
- password no
2003 Nov 10
1
Problem in MySql-3.23.49
Hi
I am a user of Asterisk-0.5.0. I am a final year student of MCA in IGNOU.All
the system are running in Red Hat-7.3 OS. I am able to transfered call in
the following procedures:
PSTN(INDIA)>>Mediatrix 1204>>Asterisk server>> VOCAL server>>Mediatrix
1204>>PSTN(USA)
Now I want to save the cdr data in my Asterisk box.I am using RedHat-7.3 OS.
I am using the
2003 Nov 12
2
Media Negotiation Failed
Hi, I have this scenario
Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip:
64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than
* )
When a calls comes in Cisco 5300, this send this calls with SIP to *,
asterisk plays a welcome message and resend call to Cisco 3600 that have
4 analog lines connected... but after cisco play welcome message and
when