Displaying 20 results from an estimated 40000 matches similar to: "Unable to create channel of type 'Zap'"
2003 Jul 16
8
Call Pickup
Hi,
I have been trying to workout how to use the call pickup.
So Far, I have the following in zapata.conf
[channels]
signalling => fxo_ks
context => local
pickupgroup=1
callgroup=1
channel => 1-3
When I dial *8# all I hear is busy tone.
What have I missed?
thanks
Jay.
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI> sip show peers
Name/username Host Mask Port Status
2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000 192.168.22.198 (D)
2005 Mar 10
2
Cisco and Asterisk
Hey all,
I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get
a bit of help here.
First I'll explain my setup, and then my problem.
Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO
ports. I have an analog phone line plugged into the first port
(voice-port 1/0/0). I've got it setup so that calls coming into that
analog line are
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
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
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 Jan 05
5
Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and
are using the CVS version.
Goto the wiki and read the instructions for RealTime.
-Matthew
----- Original Message -----
From: "Muhammad Rizwan Khan" <rizwan@advcomm.net>
To: <Asterisk-Dev@lists.digium.com>
Sent: Wednesday, January 05, 2005 12:42 PM
Subject: [Asterisk-Dev] Asterisk with MySQL
>
2005 Aug 27
2
Problems with registration
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default ; Default context for incoming calls
port=5060 ;added
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ;
2004 Jul 27
5
sip over h323
Hi List,
we are using openh323 gatekeeper for voip telefony. We also have a voip
over ss7 TELES Switch for voip into POSTN Network. Know we want to use
Asterisk for converting SIP to h323.
Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do
we need openh323 GK for astrerisk, too?. And how can i tell asterisk
to sent all none SIP-ip calls to the gatekeeper over h323?
thx
2004 Jul 26
6
Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi,
I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've downloaded
the
2005 Jul 25
3
Zap channel configuration problem
Hi,
I would like to use a digum card to call an external number through my
PSTN. I think that I have a problem in the configuration. Asterisk
returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap'
I use Fedora core 3.
I installed libpri, zaptel and asterisk.
I plugged my line on the FXS module (green part).
I make modprobe zaptel && modprobe wctdm without
2004 May 24
2
Newbie extensions.conf I need to include [SMS] context.
I want to include a new context in my exensions.conf
I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
and I can sort of follow it?!
I have a context [local] that I know zapata.conf points to, I have edited
extensions.conf and put in my phone, sip and iax extensions. I want to add
an sms context.
I understand that all calls go through my [local] context and I have
2004 Jan 29
4
dialing wrong numbers
hi,
I am new to * and setting up a test system.
here my setup :
- debian (from knoppix 3.3)
- Asterisk 0.7.1 (from the debian package)
- AVM Fritz card used with i4l
- softphone I use for testing SJphone on windows
- I can make great softphone - softphone calls
- I can call from an outside line * and get connected to a softphone
here my problem:
I can not make outbound calls. I place a call
2004 Jul 18
18
Polycom IP 500 Voicemail
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center prompts. I have followed all
the Admin Guide instructions regarding the phones .cfg files and using
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2004 Dec 13
1
Repost: Cisco 7960 and Asterisk...not working....
Anyone help me here? I am a newbie so be gentle ;-)......
It worked once and then I played with the configs.
I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest is )
I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here
2006 Mar 24
5
GSM/DECT handsets (was gsm picocells)
Now that I actually try and google for it, I can't find any dual mode
GSM/DECT handsets, only pages telling me that they exist without any
actual information!!!
Does anyone know of any such handsets? (and even better, ones that are
available in Australia) I've searched a few of the major gsm
manufacturers (nokia, Panasonic, sonyericsson) but their web sites are
absolutely pathetic to the
2004 Sep 14
1
Requested device 'ttyI1' does not exist
Hello List!
I finally got asterisk with capi working, and its already answering my
call as well! :)
Now i would like to call a number from my shoft phone (kphone).
This is my extentions.conf:
---
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
2004 Aug 10
11
CAPI call transfer
Hi,
I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI. Then
I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.
The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the
2004 Aug 20
3
Strange problem with Dial
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two different actions by *.
Here is the vvverbose output:
-- Starting simple switch on