Zaptel / Asterisk on Solaris

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Hello nice people :) I have been struggling with trying to get Zaptel from
http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained
from the OpenSolaris Website. I have tried installing all the necessary
packages, yet I keep getting errors no matter if I try using the gcc
available at sunfreeware.com OR the blastwave CSWgcc packages and GNU
'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon,
from Slimey.org who built/created this Zaptel Solaris Port, but it's been
over 2 weeks…

Asterisk Users 4.7 years ago 10 Answers

Asterisk with MySQL Cluster

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I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime I am
only able to point to a single IP address for a database. If that database
server goes down that Asterisk is pointed to then Asterisk won't be able to
do anything. Any options within Asterisk 1.8 to make it more fault tolerant
when it comes to Realtime and databases?

Asterisk Users 4.7 years ago 0 Answers

No such file or directory

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Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so
first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib /lib/ asterisk-cli>modules load res_jabber.so
asterisk-cli>modules load chan_gtalk.so
Cheers!
*José Pablo Méndez
*********

Asterisk Users 4.7 years ago 0 Answers

Rhino Channelbank...

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I am having a problem with a Rhino channelbank. I have an Asterisk
server running 1.6.2.9 and DAHDI 2.4.0 on a CentOS 5.5 system. We have
a TE420 card with the first port used in E1 mode (R2, 20 channels) and
the fourth is in T1 mode for the channelbank. We are using MG2 echo
cancellation. The Rhino unit has 12 FXO ports. Five are GSM adapters and another
five are regular phone lines. We are having problems with the GSM lines
at the moment. When we enable MG2 on those lines we get…

Asterisk Users 4.7 years ago 0 Answers

Correct operation of timout parameter for dial application

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Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial would timeout on the signalling prior to the timeout parameter specified in the dial parameter. For example, consider the following dialplan: exten => 111,1),Dial(SIP/phone1,30,tg)
exten => 111,n,NoOp(DialStatus=${DIALSTATUS})
exten => 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)

Asterisk Users 4.7 years ago 1 Answer

Default From and Contact header domain

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Hello all, I have a server which is sending INVITEs with a From and Contact header that
contains a domain part of the address (an IP address) that I can't explain.
My sip.conf does not set a domain.
For example in the following line the 123.456.789.012 is the part I can't
explain. From: "" ;tag=aa00104d30 Does anyone know where Asterisk gets the default for these headers from? Thanks!

Asterisk Users 4.7 years ago 0 Answers

Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)

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On 30 Nov 2010, at 09:47, Michael Nausch wrote:
> I tried to configure Asterisk 1.8 on one of my test-hosts.
>
> I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/):
>
> [Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No channel type registered for 'mISDN'
> [Nov 30 10:35:53] WARNING[7281]: app_dial.c:2030 dial_exec_full: Unable to create channel of type 'mISDN' (cause 66 - Channel not implemented)
>
> Is there no misdn-support activeted in the latest version, cause if I use the help-command on Asterisk's command-line-interface, I can't see misdn?! You installed the module, but…

Asterisk Users 4.7 years ago 0 Answers

TCP port, VPN and resolving the cutting voice problem

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Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing if we used the VPN, and in case of other users are using the Internet to…

Asterisk Users 4.7 years ago 19 Answers