[Asterisk-Users] Some questions (maybe Nikotel related)

[Christian Peter]

Hi  Michael,

thanks for you answer. Comments below.

Am Donnerstag, den 13.01.2005, 12:38 +0100 schrieb michael koehler:
> inline
> 
> On Jan 10, 2005, at 10:12 PM, Christian Peter wrote:
> >
> > - If I call outside (with Nikotel to German Telekom) there is a remote
> > hangup after 2 minutes. I've seen other people posting this but nothing
> > helped. I luckily managed to get around this issue with the following
> > workaround: The provider section should only contain disallow=all and
> > then only allow=gsm. If I add allow=alaw .....
> 
> After 60 seconds nikotel send a reINVITE to your box. If your box does 
> not respond
> then the call gets cleared after 120 seconds. I do not know why this is 
> up to the codec
> order of *
> >

I've got more codec problems. See below.

> > - I sniffed the traffic and came to another strange issue. From time to
> > time asterisk sends a OPTIONS packet (even before REGISTER). This
> 
> Seems that * keeps routers WAN port this way
> 
> > packets have a From header which looks like this:
> > <sip:asterisk at 192.168.1.170>
> > Nikotel does of course not recognize this address and sends a "Call leg
> > or transaction does not exist". Is this a bug or intended behaviour?
> 
> Looks like the OPTIONS request happen outside of an dialog.

Ok I forgot to give externip=xyz and srvlookup=yes in the sip.conf. Now
it uses the right ip address but still asterisk as username.

> 
> >
> > - No internal Nikotel call (phone number beginning with 99) reaches my
> > friends (which have similar sip.conf and extensions.conf). Somewhere I
> > read that the section must be named like the host 
> > "calamar0.nikotel.com"
> > so that asterisk finds it. It didn't help. Did someone manage to get
> > this working?
> 
> There is(should be) a 302 Response fix in the current CVS
> 

I tried it with current 1.0 CVS (and 1.0.2 and 1.0.3 :)). The redirect
now works if one specifies the ip of calamar0.nikotel.com as section
name. Now there are two possibilities after the phone accepts the call:

- If I remove the disallow=all, allow=xyz stuff from sip.conf asterisk
gives no error message but i can't hear something (Nikotel testnumber:
999 900 900 900)
- If I put disallow=all, allow=gsm in the sip.conf asterisk tells me
that the codecs are incompatible.

I'm now giving it up. I spent almost a week on the 99er issue and still
no luck. If someone has Nikotel FULL working I would be glad to see
his/her configs.

Greetings

Christian

> 
> Michael
> 
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