if you check out your sip.conf.
On Jun 29, 2012, at 5:54 PM, gincantalupo wrote:
> Hi all,
> after upgrading my Asterisk 184.108.40.206 to 220.127.116.11 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150.
> The strange thing is that the rport inside SIP packets ("sip set debug") coming back from my provider is set to 55150.....seen on both Asterisk 1.4 and 1.8
> Does anybody…
i have configured properly asterisk. At the one end i am using x-lite soft
ph and another end twinkle. call is going properly from both end but after
picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always showing
closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION
5060/tcp closed sip telnet localhost 5060 (could not connect)
Actually I can start and receive SIP calls (PC client, iphone client)
but I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2). I'm having this error when making a call: *CLI> == Using SIP RTP CoS mark 5
-- Executing [9000@local:1] Dial("SIP/3000-00000006",
"DAHDI/1/4384019357,10") in new stack
[Jun 23 16:18:09] WARNING: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9000@local:2] Hangup("SIP/3000-00000006", "") in new stack
== Spawn extension (local,…
Is this http://www.voip-info.org/wiki/view/Asterisk+vzaphfc page data
still up to date ?
In other words, is it possible to use One port BRI cards with Dahdi ? Regards
After an upgrade, I discovered yesterday strange things I would like
to share here. Basically, I'me comparing platforms:
The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk
18.104.22.168, Libpri 22.214.171.124, Dahdi revision 8853 (must be between 2.3
and 2.5, I think).
The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk
10.5.1, Libpri 1.4.12, Dahdi 2.6.1.
Both are connected to telco BRI lines in TE/PtmP mode through a
Junghanns QuadBRI board (wcb4xxp driver).
Both handle incoming and outgoing calls correctly, as far as I can tell. But…
i am trying to install the just from git cloned app_swift version.
Compiling works fine. Install as well. But if i try to load the module
at Asterisk it fails with. Command 'module load app_swift.so ' failed.
[Jun 20 11:29:51] WARNING: loader.c:460 load_dynamic_module:
Error loading module 'app_swift.so':
/usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close
[Jun 20 11:29:51] WARNING: loader.c:850 load_resource: Module
'app_swift.so' could not be loaded. My System Informations: server*CLI> core show version
Asterisk 126.96.36.199 built by root @ server on a x86_64 running Linux on
2012-06-20 08:55:14 UTC root@server:~# uname -r
does someone use the USB-Stick "Huawei K3765-HV" with Asterisk?
I have an 4-Port TT-Hub (Cypress Chip) and 4 USB-Sticks with 4 different
GSM Provider (Germany, France, Turkey, and Iran) Currently I use the "Vodafone Easybox 803 A" together with an ISDN
connection to my Asterisk Server, an analog telephone (PA710) and an USR
Sportster Vi 14.400 Fax-Modem (HylaFax). Grmpf, I was not abele to send faxes from my workstations direktly using
Asterisk And then the last question: Does someone know, how the OpenSMS API is working? The EasyBox 803A does support it and I…
Seeing as this an issue related to faxing using the SpanDSP library;
if you do not get an answer leading to a solution here, then you may try
asking on the SpanDSP mailing list
http://lists.soft-switch.org/mailman/listinfo It's likely that the Asterisk users, specifically using SpanDSP, may be
on that list. Thanks, Rusty Newton
Open Source Community Support Manager
Digium, Inc | www.digium.com | www.asterisk.org
On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote:
> I´m with asterisk 188.8.131.52
How I can increase the voice volume in the analogue line (at dahdi port) without getting a problems in the entered digits (when using Background function)?
I got to know previously that it is possible to increase the gain at hardware and not software, this is better to avoid getting a problems. But where? I am using DAHDI 2.4 and another machine has DAHDI 2.6
Using txgain and rxgain from chan_dahdi.conf will not help, it will increase the voice volume but with the following problems:
1) Suddenly the call will be disconnected while we are talking.
Recently I had to change the port Asterisk listens to
(non-standard, to hide from bruteforce attacks), but at the same time I
wanted to not break the system for all current users. So I needed some
way to listen to two ports for some time. I did some research in the
Internet and found the only one solution - via iptables REDIRECT For
some reason it was not working for me, and I found many discussions
saying that lots of people can't get it working either. Despite the
statistics for rule say…