asterisk users - Jan 2005 - Some questions (maybe Nikotel related)

Hi list,

I have some nontrivial questions. I am no telecommunication guru and I
will explain it with my simple words. I hope someone can help me with
these issues (with Asterisk 1.0.3):

- 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 in the provider section then
the remote hangup occurs. Thats funny because it shouldn't matter
because gsm comes before alaw in my list. Since Asterisk 1.0.3 the codec
order in the telephone and provider sections should be respected.
Someone experienced this behaviour too? Is there a developer who can
comment?

- I sniffed the traffic and came to another strange issue. From time to
time asterisk sends a OPTIONS packet (even before REGISTER). This
packets have a From header which looks like this:
<sip:asterisk@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?

- 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?

Attached are my sip.conf and my extensions.conf

Greetings

Christian Peter

##################################
SIP CONF:

[general]
port=5060
bindaddr=0.0.0.0
context=default
videosupport=no
useragent=TEST
;externip=xxx.xxx.xxx.xxx
disallow=all
allow=gsm
allow=alaw
allow=g729
allow=g723.1
allow=g726
allow=ilbc
allow=speex
allow=ulaw

register=myusername:password@calamar0.nikotel.com/calamar0.nikotel.com

[calamar0.nikotel.com]
secret=password
username=myusername
fromuser=myusername
fromdomain=nikotel.com
qualify=yes
type=friend
context=waehlplan_eingehend_nikotel
host=calamar0.nikotel.com
nat=yes
canreinvite=yes
canredirect=no
promiscredir=yes
insecure=very
;incominglimit=1
;restrictcid=yes
callerid=myusername@calamar0.nikotel.com
disallow=all
allow=gsm


[10]
type=friend
username=10
secret=xx
host=dynamic
dtmfmode=rfc2833
nat=no
qualify=1000
mailbox=10@10
callgroup=2
pickupgroup=2
disallow=all
allow=gsm

#######################################
EXTENSIONS.CONF


[general]
static=yes
writeprotect=no

[default]
include => 10

exten => _49.,1,Dial(SIP/${EXTEN}@calamar0.nikotel.com,60,r)
exten => _99.,1,Dial(SIP/${EXTEN}@calamar0.nikotel.com,60,r)

[waehlplan_eingehend_nikotel]
exten => _.,1,SetMusicOnHold,default
exten => _.,2,Dial(SIP/10,60,Ttm))
exten => _.,3,VoiceMail2(u10@10)

[10]
exten => 10,1,Dial(SIP/10,45)
exten => 10,2,VoiceMail2(u10@10)
exten => 10,3,Hangup

Hi!
> - 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?
Use "sip debug" to turn on debugging and you'll see why: Nikotel
uses
"302 temporarily moved" trying to redirect you, but Asterisk
doesn't
really like that. More in this mailing list's archives, just search on 
"302 temporarily moved"...

Also read this:

http://www.voip-info.org/tiki-
index.php?page=Asterisk%20How%20to%20connect%20to%20Nikotel

Cheers, Philipp

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 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@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.
>
> - 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


Michael