search for: chanavail

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

...o suppress outbound caller ID. I am sure I'm not the only one out there that has had this problem. How do I get around the issue? Do I need to detect the outbound span before dialing on the group, or do I need to separate the two spans into different groups and fail-over manually using a "chanavail" strategy? Ideally I'd like to just issue a single Dial(Zap/g2/<prefix><number) but I am sure my second telco will come back with a different prefix than the other. Any hints gratefull received. cheers, Mark. http://www.switchnet.com.au http://www.hearmymessage.com ----------...

Hi, Does anyone have any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? Thanks Robert McNaught -------------- next part -------------- An HTML attachment was

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a "Page". From what I understand, the "Page" is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller