[asterisk-users] SIP v IAX2

[Douglas Garstang]

> -----Original Message-----
> From: Olle E Johansson [mailto:oej at edvina.net]
> Sent: Thursday, October 26, 2006 1:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP v IAX2
> 
> 
> 
> 26 okt 2006 kl. 18.57 skrev Douglas Garstang:
> 
> >> -----Original Message-----
> >> From: Dave Cotton [mailto:dcotton at linuxautrement.com]
> >> Sent: Thursday, October 26, 2006 10:21 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] SIP v IAX2
> >>
> >>
> >> On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
> >>> with SIP qualify, I can specify, what time in delay I will accept,
> >>> with sip and setting qualify=3000 I can circumvent this
> >> anoying messages
> >>> (bacause delay in reply is about 2000ms, and I accept 3000ms)
> >>> with iax, qualify is working different, so setting
> >> qualify=3000 will
> >>> "ping" peer every 3s,
> >>> quite inconsistent, imho
> >>
> >> So are you saying that in your world two different things, 
> created by
> >> totally different people, must have the same configuration 
> settings.
> >
> > - You will find DUNDi configuration a lot easier with IAX, 
> although  
> > you can use SIP.
> > - If you use SIP to route calls between Asterisk boxes, you will  
> > lose your caller id as SIP uses the From: number to authenitcate  
> > with. You will have to store the original caller id in an 
> extra SIP  
> > header, and then pluck it out an the other end, if you want to  
> > preserve caller id. Yuck. IAX doesn't have this problem.
> 
> That is a configuration issue, not a protocol issue. You don't have  
> to configure Asterisk that way.

If we configured it, such that a call coming into an Asterisk box was handled in the same manner as a call coming in from our PSTN gateway, then we would not be able to determine which was which. Right now that isn't a problem, but I have a strong feeling that it will lead to problems in future.

Doug.