On Monday 06 Feb 2012, John Cahill wrote:
> logger -s “checksetexternip.sh: External IP address
> has changed, changing /etc/asterisk/sip_general_custom.conf” grep -v
> “externip” /etc/asterisk/sip_general_custom.conf >
> /etc/asterisk/sip_general_custom.conf.tmp echo “externip=$EXTERNIP”
> >> /etc/asterisk/sip_general_custom.conf.tmp cp
> /etc/asterisk/sip_general_custom.conf.tmp
> /etc/asterisk/sip_general_custom.conf rm
> /etc/asterisk/sip_general_custom.conf.tmp

You could also do something like:

sed -i -e “s/^externip *=.*/externip = $EXTERNIP/”
/etc/asterisk/sip.conf

Apologies for the wrapped code.

Regards,

Hi,
 
I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager).
The connection seem to be there. When I do a “sip show peers” on the Asterisk server I see the Callmanager as Monitored and online however I can’t get any calls to pass from the CM to the Asterisk. If I debug the SIP I get a regular “SIP/2.0 400 Bad Request – ‘Malformed/Missing URL’” which is from the Callmanager.
Can anybody tell me the cause of this message and/or how I can resolve the problem ?
Any help would be greatly appreciated
 
Many thanks
 
Nigel

Hi,

When dialing multiple numbers, we get connected to the first channel that answers and others are hung up.
exten => 100,1,Dial(SIP/provider/14153456789&SIP/provider/15606305670)

How can I find out the channel (actually which destination no) the connection is made to (so that I can write it to CDR)?

Thanks,
Matt

Created this function on one of my machines today, thought others might
find it useful:

freesip() {
comm -2 < (seq $2 $3) <(cat $1 | grep ^\[ | sort | uniq | tr -d [ | tr -d
]) | grep ^[[:digit:]]
}

On RedHat/CentOS based systems you can create the following file to have
the function available on login:

/etc/profile.d/freesip.sh
# Free SIP extensions
freesip() {
comm -2 < (seq $2 $3) <(cat $1 | grep ^\[ | sort | uniq | tr -d [ | tr
-d ]) | grep ^[[:digit:]]
}

Then if you have a large sip.conf and want to find a free extension between
200 and 299 you can login and do the following:

[host]$ freesip /etc/asterisk/sip.conf 200 299
200
201
233
245
[host]$

Hello guys.

When I am trying to send fax through T38 to linksys SPA (properly
configured etc. – I have tried it with other systems), I’m getting error
and fax is not delivered.

I’m getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining
ERROR[687] chan_sip.c: UDPTL creation failed
WARNING[687] udptl.c: No UDPTL ports remaining

then, couple lines down:
WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400
udptl t38
WARNING[3514] chan_sip.c: Failing due to no acceptable offer found

sip_general_custom.conf contains t38pt_udptl=yes

udptl.conf contains:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy

Asterisk version is 1.8.5.0

When I restart asterisk, everything is working good. Then, after some
time, fax stop working.
Do you have any idea what it could be?

Thanks in advance.