Displaying 20 results from an estimated 2000 matches similar to: "Can't dial SIP<->EuroISDN (HFC-S based PCIISDN card): Unable to create channel of type 'Zap'"
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 Aug 05
0
call outside from FXS through FXO
Hi,
I am trying to make an outbound call from phone attached to FXS port.
My telephone (VoIP) line is connected to FXO port (Zap/4)
Default context for channel # 4 is 'directdial'
here is part of my extension.conf
[directdial]
ignorepat => 9
exten => 9,1,Dial,Zap/4/
exten => 9,2,Congestion
include => international
[international]
ignorepat => 9
exten =>
2005 Aug 17
1
Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below
I go off hook
2005 Aug 15
2
No translator path exists for channel type MGCP & Comfort noise support incomplete
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
2005 Feb 18
0
Time to beg on my knees for help!!!
Specs: Fedora Core 3. Dual P3 600 (Dell PEdge 1300) SCSI Disks
1x X100P (channel 1)
1x TDM20 (channels 2+3)
1x Knockoff X100P (channel 4)
I am looking to have all local and all toll free calls go outbound through
the Copper line, and all long-distance and international to go out through
the Vonage line. This way I can eliminate LD on my home line, and pay
minimal LD charges through
2003 May 27
1
Duplicate numbers with outbounding calls
I've a problem with my X100P card.
I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I
call an PSTN phone number, some digits are duplicated, so I'm unable to
call the right person.
Not very clear ? I'll try to do better (sorry, I'm french...)
example :
I use ohphone (with quicknet hardware), I call asterisk
(*192*168*1*204#), asterisk answers, I choose
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
2003 Sep 11
3
PROBLEM RECIVING CALLS AT FXO
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook
in strange state 6 on channel
2005 Jan 07
4
can the dialtone be changed after pressing 9?
extensions.conf has
ignorepat => 9
exten => _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
2007 Jan 29
0
Dropped call issue with IAX Trunking
Trixbox 2.2 Beta with freePBX 2.2.0rc1
I have a setup that looks something like this in ASCII art:
Teliax IAX Trunk ------+
|
V
Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+
+--------------> Lima Office Server -----+|
2004 Jun 14
1
making * more like a normal pbx (cisco ata-186)
I've done something similar at home, but made my dialplan such that I can
dial either 10 or 11 digits locally. I don't use a "throw away" digit at all.
Any 7, 10, or 11 digit call will be appropriately mangled and sent out the
PSTN / VoIP provider.
________________________________
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On
2006 Nov 06
0
help for recording
Hello ,
I want to enable recording for a few extensions. In sip.conf it is
defined as
record_out=Always
record_in=Always
under the section of extension.but it doesn't work.
Extensions are defined in the extension_additional.conf file like
exten => 10,1,Macro(exten-vm,10,10)
exten => 10,hint,SIP/10
exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL)
I can't be sure
2008 Aug 16
0
Basic outbound calling issue : a lot closer
I get congestion (same error) with
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
not dialing 1
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
dialing 1
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
dialing 9
All the same
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [9544790554 at To_Airspring:1]
2004 Dec 29
0
12 CANCEL's followed by 12 INVITE's in 5 secs
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients(using linphonec).
In a proper context, I have mentioned extensions 107 as
simputer@X.X.X.X (x.x.x.x=asterisk server ip)
Asterisk Sever-------------------------simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on
2004 Dec 27
0
Call Placing timeouts
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients.
In a proper context, I have mentioned extensions 107 as
simputer@bogus.com
Asterisk Server-------------------------simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on asterisk terminal
---------------------
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
2007 Jul 03
1
Configuring BLF or Asterisk presence/Hints feature
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me.
Regards
FArooq
**********************************************
1
**********************************************
in my
2007 Jul 05
1
Need Help in Asterisk BLF/Presence/Hints
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me.
Regards
FArooq
**********************************************
1
**********************************************
in my
2003 Dec 18
1
Different Dial tones for internal and external.
On systems even key systems it is customary to have an 'internal' dial
tone.
Since Asterisk simply ignores the 9 and keeps the tone going it is hard
to tell for some 'new users' if they can make a call.
My first idea was to change the generated dial tone via source. Then if
the user presses 9 go to a different context where I would record about
30 seconds of the normal dial
2007 Oct 03
4
Secondary Dialtone and selecting a specific line from Zap/g
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
----------------
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone,