Displaying 20 results from an estimated 700 matches similar to: "Fwd and Tollfree"
2005 Feb 01
5
IAX registration keep alives
hallo all
could anyone tell me how to get the * to send keepalive packets over a registration "trunk"
or how to increase the amount
I'm having natting issues, (the machine is siting behind 2 nat firewalls)
thanks
liaan
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2005 Feb 07
7
IAX2 Trunk Problems with NAT
Hi,
I have successfully configured an IAX trunk between 2 asterisks, calls can
go through both ways without any problems, NAT in the middle of course
(iptables)
Now , leave them for a while , and make a call from the external server ,
it doesn't go through,
Dial from the internal one, everything works fine again..
Now , it is clearly a problem in the NAT engine,
2006 Jan 28
2
RoadRunner
I use SIP over VPN with RR from TWC no problem, connect via WiFi.
According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps
down and 353Kbps up at this time (6:15pm Saturday). My laptop currently
has an X-Lite (free version) softphone with GN Netcom USB professional
contact center headsets (GN8110 USB XP adapter). We have found that the
headset makes a major difference in the quality
2005 Jan 14
2
Passing PIN Numbers
To All
If anyone can shed any light on this it would be greatly appreciated.
My phones are unable to enter pins numbers correctly when required by the party they are calling.
For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if
2003 Apr 08
3
IAXTEL Inbound, and Outbound Tollfree Changes
Last night Mark and I made some changes to the IAXTEL tollfree outbound,
and inbound access.
The inbound access number has changed to: 248-724-0700.
(This number is in Pontiac, MI Ratecenter, and is supplied by
Telesthetic LLC, a next gen phone compnay)
This number will say "Please dial your number now" at that point
you can dial your 1-700-XXX-XXXX IAXTEL number assigned.
In the
2003 May 02
1
IAX tollfree extension conf
Hi,
I recall seeing a sample extensions.conf file that allowed tollfree
calls to be routed via iaxtel to the US and the NL, but I must be going
blind, because I've scoured the list but can't find it. Can someone
send it to me if they have it? Much appreciated. Thanks!
---
Paul Cheng
M?ty?s kir?ly ut 10
H-1121 Budapest HUNGARY
paul.cheng@alum.mit.edu
mobile: +36 30 381-9311
2005 Sep 19
2
kill a .call file
Any means of killing a .call file that is in progress?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
2003 Apr 01
3
* on openmosix
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2004 Nov 24
2
call forwarding to gsm phones
Hii,
I want to forward calls from an asterisk server to a local gsm network.
I have read the wiki pages on various forums.
But the thing i want is to receive the call(Voip) from an asterisk
server then it should be forwarded to a gsm network & again to either
a gsm/ PSTN from the gsm network itself.
Please post a help.
Thanx in advance.
--
Day by Day in Every Way I'm Getting Better
2005 Jan 13
2
SMS Gateway
Does anyone know of any companies where I can interconnect with for SMS?
?
.o-------------------------------------------------------o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
2005 Aug 23
2
YAACID isn't working
Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ )
for incomming call notification on PC (and open url with callerid), but
it does not display/pop anything :-(
my config is very simple...
(yaacid is successfully registered as manager in asterisk)
thanks
PJ
* dialplan:
'953' => 1. NoOp(${CALLERID})
[pbx_config]
2005 Jan 18
1
No compatible codecs
Original Post
----------------
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone call (e.g. from iaxcomm) gives the following on the
console:
--
2006 Apr 05
2
chan_modem_i4l delay
Hi,
I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig
of memory
When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1
minutes , 2 sec after 3 minutes and so on...
After a quart hour, the delay make the conversation just
2005 Jul 10
2
SMS Handler in Asterisk
Hello all,
Recently I migrated all telephony in my house to asterisk thanks to the
Asterisk, QuadBRI which works wonderfully well. Some small tweaks to
make but that's on the long list.
On the short list is the ability to reliable send and receive SMS.
For SMS I already built a script email2sms, but sometimes the SMS
doesn't get send from some reason, the sms log then reports something
2005 Aug 23
3
Music On Hold + canreinvite=yes
For canreinvite=yes to work, I think I need to remove the t argument in
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways
stay in the middle. I don't want that, so I removed the 't' argument.
That works. Now, when two UA are calling, Asterisk gets out of the RTP
stream. However, when removing the 't' argument, the Music On Hold
doesn't work anymore
2005 Sep 20
9
HooDaHek 0.6 Released
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call?" Much head smacketh ensued, and as I made
that change for Steven, I noticed that I had the way wrong version of
2005 Sep 21
2
Submitting ISDN-MSN from a SIP-Phone
Hello,
i wonder why i didn't find a solution for this problem yet, because it
seems very common:
I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the
Asterisk-Server and then dialing the number of the SIP-Phone.
If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server
is
2005 Feb 03
3
IAX dns lookups
Hi all
Do any of you know i can force asterisk to lookup ip
addresses for peers and
trunks everytime it tries to make a call.
One of the peers has a dynamic ip and is using DynDNS
to register host. Now
i need to reload asterisk everytime i want to call it
thanks
liaan
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2005 Feb 15
2
[LLVMdev] Removing $(LLVM_SRC_ROOT)/autoconf dependensies in Stacker, llvm-java [PATCH]
On Mon, 14 Feb 2005, Reid Spencer wrote:
> isn't necessarily tied to LLVM. Anyway, lets cross that bridge when we
> get there.
Sounds good.
-Chris
--
http://nondot.org/sabre/
http://llvm.cs.uiuc.edu/
2005 Feb 04
5
IAX2 register Refresh
Hi all
I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file
thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec.
I need to get this down to 15 sec (nat /pat firewall issue)
any ideas?
thanks
Liaan