Displaying 20 results from an estimated 1000 matches similar to: "please help (reposted) - re. * connecting to a commercial call service"
2003 May 26
1
Quetsion about DISA...
Hi all,
i use the DISA app for giving the user a trunk after a authentication
through PGSQL as follows
.... auth via PGSQL
exten => s,1,DISA,no-password|test
I think the user is now in context "test" and he could dial any number if
the extension-conf in "test" is for example
exten s,1,Dial,OH323/<myip>
But if the user dial one digit the call build up
2003 Jun 06
3
SIP codecs
i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.
How do I find out which codecs are installed, and how
can I install further codecs? Any suggestions which
would be the right one?
I think hte problem is from the
2004 Sep 18
0
Quintum A800 and asterisk
I just upgrade quintum A800 with new SIP firmware
----------
Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B)
Gatekeeper Status: Mini
GK Calls Allowed: 8
Feature Bit Status: -PS/+RB/-ER
Languages allowed: 1
Serial Number: A002-00308F
Ethernet Address: 00-30-E1-00-30-8F
IP Address: 10.101.0.10
Subnet Mask: 255.255.255.0
Default Gateway: 10.101.0.1
System Software Version:
2006 Jun 25
5
FW: Asterisk Quintum A800 SIP Mode
Hello,
I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk
as followed:
[SIP_BD1]
type=peer
qualify=yes
host=192.168.0.254
disallow=all
context=from-pstn
allow=h723
and inside the quantum I change the config sip useragent to 5060. Up to this
part if I run sip show peers, I got:
asterisk1*CLI> sip show peers
Name/username????????????? Host??????????? Dyn Nat ACL
2006 Dec 12
1
AGI problema
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<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
2006 Nov 02
1
is IAX required for firewall and router?
I'm trying to understand IAX and whether or not it would solve my
difficulties:
'The primary goals for IAX were to minimize bandwidth used in media
transmissions, with particular attention drawn to control and individual
voice calls, and to provide native support for NAT (Network Address
Translation) transparency. Another goal is to be easy to use behind
firewalls.'
2003 May 13
1
beginner's question!
hi there,
I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips :
firstly on modifying the
2003 Jun 06
1
more about SIP ...
I added the line "allow G723.1" in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.
help!!
Dave
ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
Expressa
723@216.52.153.207 : Go2Call SIP gateway
-- Executing
2006 Apr 10
3
Vertical
Hi all.
I'm in the process of configuring a phone system for my family and friends.
I'm wondering if I should try to implement the "Vertical
Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the
Asterisk dialplan, or if I should delegate those functions to the various
ATA's.
For example, the Sipura SPA 2002 can handle*69 internally. On the other
2010 Feb 17
2
asterisk dahdi fax problem
Hi,
I run into a problem and I'm not shure what do I misconfigure. I've a
B410P ISDN card with bri_cpe signalling and two Openvox (A1200, A800)
cards with fxo_ks signalling, all with dahdi drivers. I can receive fax
from a public number, but I can't send fax. The CLI says it picks up the
line but no dialing. I tried the extension with an analog phone, it
works fine, I can dial
2007 Oct 03
4
Problem with mISDN and HFC-Cards in Asterisk-DomU
Hello,
I am having problems, getting my asterisk-domU to work properly. It consists
of the following components:
- Debian Etch under Xen-3.1 with a 2.6.18-kernel
- Asterisk 1.2.24
- mISDN-1.1.5
I have 2 HFC-ISDN-cards, which I pass through to the Asterisk-DomU in
permissive mode. This is working fine.
The strange problem is, that the two HFC-ISDN-cards are not beeing initialized
by the
2004 Jan 20
0
Outbound call with Go2Call
Any got experience with these?
I couldn't fint anything in any postings...
it seems they have a h.323 on voip01.go2call.com and a sip on
sip01.go2call.com
I have tried to register with some of the same as I use for nikotel, but
Asterisk does not want to register.
I've tried to use both the user name (ingvald) and the PIN code 440.... as
authentication.
---from sip.conf----
2005 Feb 20
1
PLease help: Asterisk to Quintum interconnection
My fellows,
We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that?
Your time is very much appreciated..
Cheers,
Jessie
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2006 Apr 05
6
transforming g729 files to wav files
Hello list,
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
Tofik Suleymanov
2016 Apr 24
1
Unable to start winbindd, Could not fetch our SID - did we join?
I've been searching this lists archives and using the Googles for two days
now, and keep coming across the same messages from before 2012 with the
errors I'm getting, so either I'm seeing something new, or I've missed
something stupid.
I've been following the HOWTOs here from Samba.org. In each case below, I
uninstalled the provided Samba packages and built from source.
2005 Feb 02
0
Problemas with Basic Services.
Hi Everybody,
I'm trying to make my asterisk dial a international call from a SER request of it. My ambient is like this.
[Clients]--[SER]--[Asterisk]--[Go2Call]
Client: My SIP clients.
SER: My REGISTRAR/Proxy Server
Asterisk: All other services(Voicemail,musiconhold etc) and also acting as an UAC dialing International Calls, because SER doesn't do that sending username, password and
2014 Sep 19
3
sr-iov on Intel 82576 and rhel 7 - would not work
hi everybody
a windows kvm guest would not start, process gets killed with:
Out of memory: Kill process 21984 (qemu-kvm) score 44 or
sacrifice child
I really don't know where/what I might be missing, config
seems fine, everything looks ok - I only am not sure, do I
need to first stub a SR-IOV device like regular passthrough?
I'm trying sr-iov, having one NIC left to the host and the
2005 May 28
1
Quintum Tenor AXT800!
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio i have
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for interanet no PSTN at all just only
IPphones and analog phones connected on FXS port.Is it's neccassary to
cannect with
2006 May 23
1
Quintum Tenor DX 3020 problem to register on Asterisk
Hi,
I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with
SIP. Asterisk always returns "Username/Password mismatch".
I've tried all configurations that was on the Quintum's manual, but no
success.
I've tested the same username and password with a Linksys (PAP2-NA) with the
same asterisk box, and it worked fine. Where is the problem ?
2004 Dec 15
7
VoIP Termination
Hi all.
I'm looking to change from a standard telephone line to a VoIP phone line at
home. I'm looking for recommendations for VoIP providers that I can use with
Asterisk.
One of the catches is that I often telecommute and sometimes I do some side
business; these practices violate many provider's acceptable use policies.
So, I need a provider who doesn't care how I use the