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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Asterisk question



 
 
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  #1  
Old March 27th 06, 05:30 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default Asterisk question

Hi

I currently have a PAP2, and have several different SIP providers (some
good/some not so good), as your aware the PAP2, can only have a single sip
account per line. To get all of these providers together on one single
account, i was thinking of installing asterisk on a colocated server and
somehow get that to manage everything for me (i'll look into the finer
details later on).

Can anybody confirm that what im going to try is possible?
Would running it from a colocated server (about 20ms away according to
windows ping), cause any bad effects?
If i make a SIP call to PSTN, will it correctly send the CID of the sip gate
provider i have setup in the dial plan.
If i get an incomming call on any of the SIP providers dial in numbers, will
the asterisk send my PAP2 (and my phone) the proper caller id of the
originating phone call?
Is there any disadvantage to doing it this way? Please let me know.

At the moment im fairly new to SIP, but im pretty good with computer
software/hardware, but would like some little advice if what im doing is
possible, rather than spending a lot of time on something which isnt.

thanx for any advice


  #2  
Old March 27th 06, 06:07 PM posted to uk.telecom.voip
Thomas Kenyon
external usenet poster
 
Posts: 254
Default Asterisk question

lab wrote:
Hi

I currently have a PAP2, and have several different SIP providers (some
good/some not so good), as your aware the PAP2, can only have a single sip
account per line. To get all of these providers together on one single
account, i was thinking of installing asterisk on a colocated server and
somehow get that to manage everything for me (i'll look into the finer
details later on).

Can anybody confirm that what im going to try is possible?


Certainly possible, could be tricky to secure.

Would running it from a colocated server (about 20ms away according to
windows ping), cause any bad effects?


Yeah, you may get a vast bandwidth bill.

If i make a SIP call to PSTN, will it correctly send the CID of the sip gate
provider i have setup in the dial plan.


That depends on the provider, some providers let you add numbers and
send that as your callerID (doesn't appear to work with voip.co.uk).

Most providers seem to force the outgoing caller-ID to match the account
you terminated the call from.

If i get an incomming call on any of the SIP providers dial in numbers, will
the asterisk send my PAP2 (and my phone) the proper caller id of the
originating phone call?


This is the default setting, you can do lots of things in asterisk to
change this behavior (change it to number then a note to say which
number they called for instance) but so long as you don't tell it to do
something else, this is how it will behave.

Is there any disadvantage to doing it this way? Please let me know.

Apart from redundancy (adding an extra stage to the calls), and the
bandwidth cost. (when you can simply get a different ATA).

At the moment im fairly new to SIP, but im pretty good with computer
software/hardware, but would like some little advice if what im doing is
possible, rather than spending a lot of time on something which isnt.

Is very possible, just try to make sure it's absolutely what you want to do.

Do you have a good grounding in POSIX-based systems?

thanx for any advice



  #3  
Old March 27th 06, 07:48 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default Asterisk question

Is very possible, just try to make sure it's absolutely what you want to
do.


Thomas, i really appreciate the reply, thanks!!! I think you have talked me
around to doing the right thing, which is either swap my ATA, or find a
single SIP provider who has everything i want, again many thanx!

Whilst im posting, i find the call quality of my sipgate account to be
great, however my sipphone account seems to be having about a 2 second
delay, considering all the options in my PAP2 are the same (except account
and the server address) im guessing this is a sipphone issue? Does this
happen to all sip servers that are located in the states or is this just a
bad provider? Call quality on both of them is excellent, but sipphone seems
to offer better features.


 




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