[Asterisk-Users] agi problem

2005-05-03 Thread M.N.A.Smadi
hi; am using an agi script to do some call forwarding. I got the following pyton script off http://home.cogeco.ca/~camstuff/agi.html: #!/usr/bin/python import sys,string class AGI: Class AGI facilitates writing AGI scripts in Python. Exported functions:

[Asterisk-Users] SIP messagse

2005-03-23 Thread M.N.A.Smadi
hi; say i have two users A and B registered with asterisk. A sends an INVITE to B thru *. My question is how can i re-write some of the parameters in the SIP or SDP message sent from A to B? thanks m.smadi ___ Asterisk-Users mailing list

[Asterisk-Users] digium card

2005-03-11 Thread M.N.A.Smadi
hi; does any body know what are the physical dimension of a digium care 400pm for example? thanks m.smadi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

Re: [Asterisk-Users] Wireless LANs and Asterisk

2005-02-10 Thread M.N.A.Smadi
what aspect exactly are you talking about? VoIP capacity over WLANs, codecs, delay, what? mohammed smadi Colin Anderson wrote: Has anyone had any experience with wireless LANs and Asterisk? I have played with the LocustWorld distro but not at length. Basically, it works. Some sort of QoS

[Asterisk-Users] logging events with time stamps

2005-02-09 Thread M.N.A.Smadi
i want the to find out the delay between two events: 1) the instance a call is recieved on an FXO port and the 2) the instance a SIP INVITE is sent to the SIP destination. i need to attach timestamps to the events before logging them. How can i: 1) log `ALL' events. 2) Attach timestamps to them?

Re: [Asterisk-Users] SIP with Delay

2005-02-09 Thread M.N.A.Smadi
no this is not normal and the codec has very slight effect on the delay. Delay is a function of two things: 1) Transcoding 2) Routing (dialplan routing + network latency) Network latency Transcoding+dialplan routing. If you are using two sip client which are using the same codec then no

Re: [Asterisk-Users] volume too low.

2005-02-09 Thread M.N.A.Smadi
it probably has to do with the handset of your phone. Try using a good mp3 from the web and test it using that mohammed Ousmane Doukara wrote: Est ce que quelqu'un peut me dire pourquoi le volume de mes enregistrement sont tout le temps trop bas. Hi, I am trying to figure out why my recorded

Re: [Asterisk-Users] Asterisk problems behind firewall

2005-02-09 Thread M.N.A.Smadi
1 2000 UDP is wrong try 1-2UDP and try port forwarding rather than opening ports . It probably has to do with the ip of the server. mohammed Jeff R Glassman wrote: I had my [EMAIL PROTECTED] server working fine SPA-8841 SPA-2100. It was on an open IP no fire wall. I moved the

[Asterisk-Users] conference room capacity question

2005-01-30 Thread M.N.A.Smadi
hi; have couple of questions regarding meet_me conference room application: 1) is there a maximum allowable number of concurrently active conference rooms per server? 2) what is the maximum allowable number of users in a given conference room before quailty creeps out? thanks moe smadi