Displaying 20 results from an estimated 900 matches similar to: "Calling an outside phone number as part of a hunt"
2005 Feb 24
2
softphone has problem to call out via X100P card
Hi all,
I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC.
With the following configuration, I can use one softphone (2000) to call the
other one (2001) and/or the voicemail at 2999.
Here is my problem:
1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via
X100P card, I got busy tone. (i.e. I want to use the phone line which is
connected to the
2003 Jul 28
1
Call Forwarding and DND conf
I have put together this call forwarding and dnd config:
I'm sure it can be dome with macro's but I couldn't figure that out...
anyone care to input.
74 Turns DND on my phone will not ring, drops caller to voicemail...
73 Turns DND off
72+ext forward your extension to another extension and voicemail is left
at the forwarded extension.
71 turns off call forwarding.
; dnd Could
2005 Jul 05
4
Asterisk on Linksys WRT54G
Hi all,
Any one tried installing Asterisk on Linksys WRT54G? We have but facing
problems with SIP to SIP calls. The phones ring and calls are established
but we cannot hear any voice at all. I tried allow=all in the general
section but did not work. So I forced ulaw. Can any one please check it out
and let me know what is wrong?
Here are the conf files:
Asterisk Version: Asterisk
2003 Nov 15
2
Internal server error - cannot align media streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different
ip softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The call from one phone to another does get routed via asterisk, but
there is one problem coming up. As soon as call is accepted by the end
user , it is automatically disconnected with the error "cannot align
media streams". If I enable SIP
2004 May 04
3
g.729 - licenses and opinions
I have a few SIP phones, Cisco 7960s, and was looking into implementing
some compression, ala G.729. I'm looking into purchasing a g729
licenses just to get an idea of performance and voice quality, over
lans, wireless and single channel isdn.
Does anyone have positive/negative experience w/ getting
licenses/support from Digium? Hows the sound quality compared w/
g.711? Is 729 better
2005 Mar 18
1
Broadvoice hangs-up / disconnects after about 30 deconds
I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message:
Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22'
-- Got SIP response 404 "Not Found"
2004 May 20
4
x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as channel 1 and
zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then issues the
following errors:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at
2005 Mar 18
2
No sound when calling in from pstn
I am just starting out with * so bear with me please.
I have tdm400p with 4 fxo modules on it. When I call into the asterisk
box from my mobile, I can see the asterisk console picks the call up
and routes it to my computer with x-lite. There was no sound coming
from either - just silence. I then decided to route it directly to
voice mail to see if that would narrow the problem down, but it
2005 Aug 17
4
Voicemail Retrival
Hi,
I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions.
Any ideas??
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2005 Jul 01
1
asterisk newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)???
are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering...
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa
Sent: Fri 7/1/2005 6:43 PM
2003 Oct 11
1
SIP / IAX over satellite
Hi all,
------
I tried to use * over satellite, but all my effort did not succeed.
The Asterisk is behind the VSAT and is resposibel for alle the SIP
clients in a field location.
The clients are notebooks and PDA's running SJPhoen for Windows and
PocketPC. Unfortunately
I could not find any Linux Client wich worked satisfying. SJ LAbs
promised a Linux Version at the end of
August but they
2005 Feb 10
4
why asterisk is replying 404 Not Found
[3000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
[2000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
i have declared these two users 3000 and 2000. they
are registering successfully.
problem is that
2003 Nov 02
6
Asterisk behind LinkSys NAT Routing
Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside
2004 Jun 23
1
Asterisk user/host registration
Hi Folks,
I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server.
When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below.
*CLI> sip show peers
Name/username Host
2004 Jul 05
4
Question about x100P and zap
I have 2 X100P card and configured everything based on configs here
http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to create channel of type 'Zap'
A zap show channels gives me this
Chan Extension Context Language MusicOnHold
1
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2005 Aug 10
1
asterisk query mysql problem or bug?
Hi;
I have entries as below in DB,
mysql> select * from sip_buddies;
+----+------+----------+------------+---------+------------+--------+-------
-----+------------+----------+------+
| id | name | context | defaultip | host | mailbox | type |
regseconds | ipaddr | username | port |
+----+------+----------+------------+---------+------------+--------+-------
2007 Dec 09
2
Don't enter a queue if no one is logged in
I currently have the following setup:
exten => 2000,1,Playback(/var/lib/asterisk/sounds/Greeting)
exten => 2000,2,Queue(Qabcdef|t)
exten => 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy)
exten => 2000,4,Hangup
exten => 2000,103,Hangup
What happens is, that the greeting in step one is played regardless if
anyone is logged into the queue. So immediately after the greet, we
2005 Jan 30
4
Processing incoming calls with multiple contextst over PRI
So I have a problem. A customer of mine wants a PBX, owns an office
building. I want to sell him on asterisk. He has 4 tenants. I am using
my asterisk box to simulate it. My asterisk box has a TDM400P card, not
a PRI card. Don't know if it makes any difference.
Anyway, I want to route incoming phone calls to different contexts based
on the phone number being called.
Here is my
2003 Oct 10
2
Grandstream Setup
Hi People,
Ok i've tried everything I can think of but cant get this to work.
Can someone please give me an example of their sip.conf settings and also the
details of the settings in their grandstream phone to allow:
1. Grandstream phone to register with asterisk when on same lan.
2. Grandstream phone to register with asterisk when phone is behind a nat.
Regards,
Aaron.