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

2006-09-27 Thread 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

[Asterisk-Users] Action: Originate PROBLEM

2006-06-22 Thread 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

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

2005-12-06 Thread 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 napisaƂ(a): lokotes wrote: When sip device sends to Asterisk INVITE with no 'Contact' field, the server

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

2005-12-05 Thread 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'.

[Asterisk-Users] PAP2 and double ringback tone

2005-11-22 Thread 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

[Asterisk-Users] SetCallerPres problem

2005-05-04 Thread 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

[Asterisk-Users] specific call transfer

2005-01-07 Thread 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. ___

[Asterisk-Users] callerid PSTN-IAX problem

2004-12-07 Thread 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. ___