* You are viewing the archive for November, 2010

Trying to configure a SIP software phone

I have been told that my logic in extentions.conf is wrong in trying to configure a SIP
software phone called Express Talk (remote) .

I’d like to make outgoing calls and calls to local extensions.

Could someone please look at my configuration files at http://pastebin.com/ajp62wqF
and see what I did wrong?

Thank you,


Zaptel / Asterisk on Solaris

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 and I’ve not heard anything from anyone. This is EXTREMELY
critical for me to work…can anyone kind generous gentleman please help?

Thank you so much

Asterisk with MySQL Cluster

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?

No such file or directory

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


*José Pablo Méndez

Rhino Channelbank…

I am having a problem with a Rhino channelbank. I have an Asterisk
server running 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

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 dropped calls
or noise or the call cannot go through. If I disable eco cancellation
it mostly works (one port still drops the call almost immediately) but
there is echo.

Here is the DAHDI configuration:

# Span 4: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″ B8ZS/ESF
# termtype: te

Here is chan_dahdi.conf
; Span 4: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″ B8ZS/ESF
signalling = fxs_ks
channel => 94-97,104-105

Does anyone have any suggestions on how to enable echo cancellation on
the Rhino Channelbank?

Correct operation of timout parameter for dial application

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=${DIALSTATUS})
exten => 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten => 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten => 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten => 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten => 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten => 111,n(busy), Goto(voice-mail,vmb-phone1,1)

Under normal operation the originating caller is passed through to voicemail. However, if/when the device is not responding to invites, for whatever reason, the dial application waits 30 seconds before setting the DIALSTATUS to NOANSWER. Is this expected behaviour? In previous versions of asterisk, specifically (v1.2/v1.4) when the device did not respond to invites the dial application exited prior to the value specified by timeout.

Can anyone clarify this issue for me please? Is this expected behaviour?

We are currently running v1.6.2.13