Hi

I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up in 1.6 However because the argument delimter in 1.6 has
changed from pipe to comma this breaks as the AUTH line is also comma
delimited. Thus the AGI sees the AUTH as extra arguments instead of a
single argument. As the AUTH may contain varying number of arguments I
need a new way for a my AGI to access this data.

Does anyone have any ideas how I might go about this?

Regards

Jon

I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.

my sip conf

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid=”Thomas Johnson”
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip[1001];Work
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=sip.domain.com
nat=no[1000];IPKall
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=voiper.ipkall.com
nat=no

I pasted the log here -> http://pastie.org/1163238

I have tried connecting both of the clients to another sip
service(sip2sip.info) and did not have the same problems.

Any suggestions would be great.

Thanks,

Tom

I have an incomming call but when I receive a call by a 2nd line in my
softphone, lost the first call. Sometimes the first call is dropped, and
sometimes the call is active, but I can’t hear the caller.
It’s an asterisk Bug? I have asterisk 1.4.22.
Please help!!!
Thanks

Greetings-

First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I’m hoping they can be of assistance.

I have a system running Asterisk 1.4.27 (see… relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID’s coming in via PRI, SIP, etc. If a user calls outbound to one of these numbers, it goes out to the PSTN (using one channel of $0.0x/min), then comes back in on another channel (using another $0.0x/min).

Obviously, the one call is costing 2x the per minute rate when it could be costing nothing. Is there a way to tell FreePBX to check the inbound routes for a match, and if found, route locally instead of using the default PSTN routes?

I appreciate any comments, suggestions.

Hello All!

I want to add a silence to the beginning of a ring tonelist for a country inside the indications.conf file. I want that silence to be played just once, reason why am using an exclamation mark in front of the tone but is not working. Am getting the ring tone right away. I tried these combinations below and some others but no help:

ring = !420*40/100,!0/5000,420*40/2000,0/4000
ring = !0/5000,420*40/2000,0/4000

Is like if the exclamation mark is not been recognized. Am not sure if it has anything to do with the fact that this is been used on a SIP channel. I am using the “r” parameter with in the dial command to generate this tone obviously.

Regards,