Displaying 20 results from an estimated 3000 matches similar to: "pattern matching problem"
2005 Jan 17
2
iaxtel - -- Format for call is ADPCM
When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf gsm
Call accepted by 69.73.19.178 (format ADPCM)
-- Format for call is ADPCM
My settings:
[general]
port=4569
register => xxxx:xxxx@iaxtel.com
bandwidth=high
jitterbuffer=no
tos=lowdelay
[voipjet]
type=peer
host= xxx.xxx.xxx.xx
secret= xxx
auth=md5
notransfer=yes
context=incoming
disallow=all ;
2005 Jan 02
1
Clipping on outbound calls via SIP/IAX
I'm hoping someone can help me with a problem I've been having for a while
now. I've googled and wiki'd to no avail.
Whenever I place an outbound call from * to a PSTN through a SIP or IAX
provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the
remote call are clipped (muted). For example, if I call a remote voicemail
system that usually answers with
2005 Jan 13
1
Teleconferencing?
I am just now investigating Asterisk. Can Asterisk provide 6-10 party
teleconferencing when configured properly?
2005 Jan 21
2
gpg key centos 3.4
Hi,
I install CenOS 3.4, how import the gpg key ?
Thx
2005 Feb 20
7
bridging iaxtel calls to PSTN
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
FSX modules) and is connected to the PSTN. B has same, but is NOT
connected to PSTN. I want to configure B to call A via iaxtel, and
connect to the PSTN using A's line. How can I configure iaxtel dial
plan for B in extensions.conf? I want to be
2003 May 05
6
IAXTEL toll-free gateway
I have been playing around with asterisk for a week or so now and
haven't had too much trouble getting things to work but one thing seems
to puzzle me. I have been patient hoping that there was a configuration
error on the server or that the toll-free gateway was down but nothing
has changed. I have the following configuration context for IAXTEL:
[iaxtel]
exten =>
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
2005 Mar 19
1
DISA -> macro = congestion
When I use DISA I get congestion when I try to reach 1-800-number:
Here is the context:
[disa]
exten => 087,1,Answer
exten => 087,2,DigitTimeout,8
exten => 087,3,ResponseTimeout,20
exten => 087,4,Authenticate(985)
exten => 087,5,DISA(951|disa-access)
[disa-access]
include => tollfree
include => outgoing-voipjet
[tollfree]
;
; terminate toll-free no.'s via fwdnet
; US
2005 Mar 11
4
VoipJet Terms of Service
I've heard good things about VoipJet here, so I was going to set up an
account. Then I noticed their Terms of Service here:
https://www.voipjet.com/tos.php
Several things there are very concerning to me, and I'm interested in
what other people here think of them.
* The ToS specifically forbids use for any call relating to medical,
financial, or government matters -- as well as any
2005 Jan 15
2
IAX2 Channels & Bandwidth
Hi all,
I'm using VOIPJET to make international calls with an IAX2 connection
between my local asterisk server and their server(s).
At times I seem to have a problem if 5 or more international calls are made
at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the
asterisk server uses this DSL line). Today I switched the codec from ulaw
to ilbc in an attempt to lower
2005 Aug 12
3
Voipjet experiment
Hi List,
I'm wondering if someone who uses VoipJet as their termination service
would do me a favor.
If I call the American Airlines reservation number (1-800-433-7300), the
call gets connected, but after 30 seconds asterisk drops the call
responding that no one answered.
I'm using areskicc2 (calling card app) as an authentication system and I
don't know if that is what is
2004 Aug 24
2
Grandstream Budgetone BT-101 and VoipJet
Is anyone using this combination successfully? I have a dell 500sc
running rh9 and asterisk 1.0rc1. It is configured with an x100p. I
have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
BT-101. I have signed up with Voipjet (they use iax2). I also have
an FWD number via iax2. I can sucessfully call back and forth to all
devices via zap, sip, and fwd. I can successfully
2006 May 23
1
Configure Voipjet.com content in Asterisk
Hi,
I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone.
Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution.
Here I am sending my configuration file values:
Contents of
2005 Jun 22
1
missing cdr records
I am experiencing a very wired problem.
Some of my cdr are lost.
I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning.
I am running asterisk 1.0.7; this is simple configuration file:
extensions.conf
[general]
static=yes
writeprotect=no
[macro-gw-voipjet]
exten =>
2004 Dec 26
7
IAX Registration Refused
I tried to connect my * to IAXtel, but i always get this errors.
chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected:
Registration Refused
On dial a iax number i get:
chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority
found
chan_iax2.c:5528 socket_read: Immediately destroying 3, having received
reject
chan_iax2.c:2411 iax2_hangup: We're hanging
2006 Mar 14
2
Max retries exceeded to host...
The past two days, I've been having issues with my two VoIP service
providers where calls just suddenly hang up. The following is from the
log:
Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host
64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=250000,
seqno=80)
Mar 14 13:50:55 DEBUG[10428] channel.c: Didn't get a frame from channel:
IAX2/voipjet-3
Mar
2007 Jan 30
2
Problem with Voipjet ...
Hello, we have this problem with Trixbox 1.23
I have created an outgoing route where the 1st line
has Voipjet and the 2nd an 3rd have voipcheap accounts.
The problem is that at certain moments, when we call all
the calls go through the voipcheap SIP accounts SIP, whose
quality are not only not good enough but also consume a lot
of bandwidth.
The error message that returns Voipjet to Asterisk is
2005 May 12
14
voipjet anyone?
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",
2005 Jul 18
4
Teliax to VoIPJet
I'm trying to setup asterisk to accept call from Teliax, request the
10 digit number from user, then dial it thru the VoIPJet. If I'm not
wrong I will be charged by both providers because both connection is
active during conversation. So my question is can I set the things so
that I pay only to VoIPJet? Specific configuration snippets will be
greatly appeciated.
Thank you.
2005 Mar 09
3
NuFone + VoIPJet = busy busy busy
Hi List,
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
Also it seems that VoIPJet takes forever to return 'circuit busy' while
NuFone does it instantly.
At any rate, is there like a reliable third VoIP provider I can use for
fallback when the two others are busy?
Cheers,
Jean-Michel.