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Summary:ASTERISK-01980: first second is truncated if call is established through an ip gateway
Reporter:chris_de (chris_de)Labels:
Date Opened:2004-07-09 10:28:27Date Closed:2011-06-07 14:10:44
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Description:calling with asterisk to an ip gateway like nikotel or sipgate the first second gets "lost" everytime. This is not the case using x-lite with an ip gateway.
This is very annoying since the first second is the time the called person usually says his name.
Its like making a capi connection without having Early-B3 enabled.

****** ADDITIONAL INFORMATION ******

extensions.conf
exten => _0X.,1,Dial(SIP/${EXTEN}@sipgate)

sip.conf
[sipgate]
type=friend
canreinvite=yes
secret=bla
username=12345
host=sipgate.de
fromuser=12345
nat=no

Comments:By: Brian West (bkw918) 2004-07-09 12:11:47

This isn't a major bug.  Its just a minor anoyance.
By: Brian West (bkw918) 2004-07-09 14:13:25

try canreinvite=no

bkw
By: Mark Spencer (markster) 2004-07-09 17:50:22

What are you calling from?  And in which direction is the audio truncated?
By: chris_de (chris_de) 2004-07-10 05:33:03

canreinvite=no doesn't work with ser (sipgate) - If I set canreinvite=no, I don't hear anything.

Also I tried
[abc]
exten => _0X.,1,SetCallerID()
exten => _0X.,2,Answer
exten => _0X.,3,Dial(SIP/${EXTEN}@sipgate)

Seems to be a little bit better, but still the first second, the "hello", is getting cropped. In this configuration I'm calling from Asterisk with public IP without any nat (from mgmt console) to sipgate twice and bridge the call.

Action: Originate
Channel: SIP/12345@sipgate
Context: abc
Exten: 012345678
Priority: 1
Callerid: 12345

PSTN <<-- sipgate <<-- asterisk -->> sipgate --> PSTN


But it's the same as if I were calling from Asterisk to sipgate or nikotel. Then the "hello" of the called person gets also truncated.

IP-PHONE --> ASTERISK --> NAT --> nikotel --> PSTN

If I only do

IP-PHONE --> NAT --> nikotel --> PSTN

it works. So I think its asterisk?
By: connor (connor) 2004-07-11 14:20:02

What version of asterisk are you using?
By: chris_de (chris_de) 2004-07-12 05:52:08

conner you are right. It seems to work with the newest cvs. => solved