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Updated configuration guides with Asterisk settings for VoiceMeUp are available for both SIP and IAX.

Digium Asterisk - Configuration Guide for SIP | English - Français
Digium Asterisk - Configuration Guide for IAX | English - Français

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Interconnect A party & B party (SS7, ISDN, VoIP)
by picaso Sat 06 of Sep, 2008
Dear Sir,

I have to interconnect A party and B party for direct route.

A party = cisco as5400
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Seeking info or advice about Call Monitoring/Recording
by heirrook Fri 05 of Sep, 2008
What I need to do is record telephone calls that are made on regular phones on a regular telephone line, and record the calls somehow using a VoIP system.

I wish to use this system using a PBX and VoIP phone adapters without using an actual VoIP external line. The line that comes into my house is a POTS line. The setup does not look like it will work to me, but I have seen this exact setup where I work so I am very confident that it does work. I wish to do it like this:

Regular Phone ----> phone line splitter -----> VoIP Phone Adapter -----> PBX
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Re: Calling Card call type CDR
by bcnit Fri 05 of Sep, 2008
There are two ways of doing this which immediately spring to mind (there will be many many more).
The first is to use the LOCAL channel and then initiate the call from within that. This should create a new CDR from that point in the call - I know we've used it for this reason, but it was a while ago, so I can't remember if we needed to do anything else.
The second way would be to log the times to the database that the dial started and when (if) the call is bridged. FUNC_ODBC is your friend!

Those should push you in the right direction, but it is worth bearing in mind that there are a number of issues with CDRs - it is not recommended that you use them as your only source of billing information unless you are absolutely sure that you are recording exactly what you want.
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Calling Card call type CDR
by sergiogal1 Thu 04 of Sep, 2008
Hi,
I wrote a DialScript that receibe a call from the PSTN, ask for destination (with a READ) , and then make a call using a Dial to the final destination.
I need to have a CDR that represent ONLY the second leg of the call (with your own billsec, and disposition, etc)
Anyone know how to do this??? HELP ME!!!!
Thanks.
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ExternIP + ExterHost Sip Parameter + DNS Resolution
by mlecuona Thu 04 of Sep, 2008
I would like be sure that when using "exterhost" parameter, although it is not recommended for production environments, the DNS lookup made to determine the IP for "exterhost" will not use the DNS server posted on /etc/resolver. Because if I am running also a name server inside my local network the result for this query will never be the public IP for the Asterisk behind the NAT.