search for: istp

Hi, My setup is : PSTN --------- ISTP Network ----------- Router ------------- Asterisk ---------- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I...

...inate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use a simple "dial" command to send it back to my ISTP using a SIP or IAX channel and the ITSP terminates it on the cell phone. One of my main goals is to keep my Asterisk box out of the media path and let the ITSP handle all the provisioning for the call. I understand that the default behaviour of the "dial" command is supposed to do jus...

...inate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use a simple "dial" command to send it back to my ISTP using a SIP or IAX channel and the ITSP terminates it on the cell phone. One of my main goals is to keep my Asterisk box out of the media path and let the ITSP handle all the provisioning for the call. I understand that the default behaviour of the "dial" command is supposed to do jus...

...here incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing calls can be made successfully and no error messages or warnings are reported by Asterisk. However, incoming calls appear to enter my dialplan as desired and go so far as to start ringing my SIP phone (Grandstream GXP-2000) but drop after two rings. The caller gets a busy tone...

Hi, I've noticed that asterisk seems to stop relaying DTMFs after a call has been up for a while (~10 mins). I was just wondering whether this was intentional, or a bug. In detail here's my setup SIP Gateway --> Asterisk --> E1 --> Asterisk --> SIP Gateway The LHS gateway sends RFC2833 DTMF messages to the first Asterisk which bridges them onto the E1. They then get

...o avoid a NAT, having SER on my Internet gateway makes most sense, which led me to Milkfish on a Linksys WRT54G. Having never worked with this before I wasn't sure how it would be configured in the end, so I just "went for it". I assumed that OpenSER would sit between Asteisk and my ISTP and handle both ends of the resgistrtion. I was able to register two Xlite phones to my WRT54G and have them talk. Not so lucky w/ Asterisk. I was able to create an account on OpenSER for my DiD and have Asterisk register on the router, but how would the ITSP end know where I am? I was hopeful...

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