[asterisk-users] my (SIP) INVITE is ignored

[lokotes]

Hi,
I'm struggling with this kind of problem:
my hardware sip phone is registering to Asterisk 1.2.10 successfully, 
but when I send INVITE to server - it receives the packet but (in sip 
debug mode) I see: 'Ignoring this INVITE request'.
While searching in 'chan_sip.c' I've found that this message shows up if 
variable 'ignore' is being set. This is when


if (p-ocseq  (p-ocseq != seqno)) {
ignore = 1;
}

unfortunately there aren't any comments around, so anyone could explain 
what exactly is happening?


regards,
L.
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[Asterisk-Users] Action: Originate PROBLEM

[lokotes]

Hi,
I'm straggling with setting up a call via manager interface. Basic 
functionality works fine but I try to use this addons:


Application: Playback
Data: beep

when a call is answered by A side, 'beep' is played correctly but no 
further action is taken - I got hangup !!!

Why it's not connecting to B side of connection after playing the 'beep'.
When removing 'Playback' all works fine.

What's the trick?

regards,
L.
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Re: [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.

[lokotes]

Thanks, your are absolutely right - I was thinking of REGISTER.
I couldn't find any information about that - if it's a known problem why 
 it's so hard to find any info?


Kevin P. Fleming napisał(a):

lokotes wrote:

When sip device sends to Asterisk INVITE with no 'Contact' field, the 
server should respond with all information it holds about client. When 
reading database fields, 'fullcontact' is empty. So, whole procedure 
ends with 'chan_sip.c:6393 register_verify: Failed to parse contact 
info'. Interesting thing, internal database (CLI databse show 
SIP/Registry x) holds all valid information about this client, so 
why it's not used?



This is completely wrong; if the SIP peer sends an INVITE with no 
Contact information, the request is invalid.


Are you talking about REGISTER? If so, that's a known problem, that 
Asterisk does not currently support registration queries.

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[Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.

[lokotes]

Hi,
When sip device sends to Asterisk INVITE with no 'Contact' field, the 
server should respond with all information it holds about client. When 
reading database fields, 'fullcontact' is empty. So, whole procedure 
ends with 'chan_sip.c:6393 register_verify: Failed to parse contact 
info'. Interesting thing, internal database (CLI databse show 
SIP/Registry x) holds all valid information about this client, so 
why it's not used?

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[Asterisk-Users] PAP2 and double ringback tone

[lokotes]

Hi,
I have a problem with double ringback tone - outgoing connections to 
PSTN. I do not use 'r' option in Dial function so I expect to hear 
'real' sounds from pstn provider. But PAP2 generates extra ringback tone 
itself! How to get rid of that?


Regards,
L
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[Asterisk-Users] SetCallerPres problem

[lokotes]

Hi,
Background:
I'm running 2x * boxes.
Box A has a registered user which dials a number. The connection is sent 
to Box B which acts as pstn gateway (sangoma 1xE1 card).

Problem:
On Box A before executing Dial() command I set 
SetCallerPres(prohib_no_screened) but despite that Box B sends the 
connection to pstn with allowed_not_screened flag ? Why is that?

When I set SetCallerPres(prohib_no_screened) on Box B it acts properly.
But why sending this flag between 2 8 boxes doesn't work for me?
Any suggestions?
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[Asterisk-Users] specific call transfer

[lokotes]

Hi,
is it possible to transfer an incomming call to another ext. without 
answering? I'm not talking about (un)conditional redirection but 
functionality, when calee can each time decide whether answer the phone 
or transfer it to any other phone.

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[Asterisk-Users] callerid PSTN-IAX problem

[lokotes]

Hi,
I cannot see cid for incomming call from PSTN (Quintum gateway) to IAX 
client (FireFly). Client displays blank but when I look into cdr's 
/var/log/asterisk/cdr-cvs/Master.cvs, the callerid is registered 
properly. Why it's not displaying?
L.
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