[Asterisk-Users] Latest CVS freakout on iconnect calls

[Mark Spencer]

Try turning off reinvites by adding:

reinvite=no

to the appropriate peer or user section.  If either peer/user is set for
no reinvites, then they will not take place.

Mark

On Sat, 15 Feb 2003, Brian Capouch wrote:

> I'm not smart enough to know much of anything more than something has
> changed drastically with the latest CVS version of asterisk as compared
> to the version I was using previously, CVS a/o 1/17/03.
>
> I am using an ATA186 to connect to iconnect, and am behind a NAT
> firewall.  I have posted previously that inbound calls to the number
> iconnect gave me have never worked, but I have enjoyed pretty much great
> quality outbound calls.
>
> I am set up so that +11 dials out through iconnect and +91 dials out
> through my default LD carrier which goes through an X100P.
>
> Calls still work fine through the X100P interface, but now my outbound
> audio through iconnect is distorting something fierce; it's a sort of
> throbbing up-and-down of the signal that sounds like it has been run
> through a phase-shifter box.
>
> There are some (perhaps) notable things showing up at the CLI screen.  I
> am getting continuous loops of messages similar to the following, with
> the integer numbers (such as 23 and 25 below) counting up from 1. They
> keep coming continuously once I start up asterisk:
>
> . . . .
> NOTICE[114696]: File chan_sip.c, Line 2728 (handle_response):
> Registration successful
> NOTICE[114696]: File chan_sip.c, Line 2729 (handle_response): Cancelling
> timeout 23
> NOTICE[114696]: File chan_sip.c, Line 1805 (transmit_register):
> Scheduled a timeout # 25
> NOTICE[114696]: File chan_sip.c, Line 2728 (handle_response):
> Registration successful
> NOTICE[114696]: File chan_sip.c, Line 2729 (handle_response): Cancelling
> timeout 25
> . . .
>
>
> I don't know if that stuff up there may be material to my problem.  The
> only other thing I can see is a difference in the way that asterisk
> spits out what's going on at the CLI interface:
>
> Previously: (Good quality outbound audio)
>
>      -- Called 666612192538552 at iconnect
>       -- SIP/213.137.73.140:5060 answered SIP/192.168.1.7:5060
>       -- Attempting native bridge of SIP/192.168.1.7:5060 and
> SIP/213.137.73.140:5060
>       -- Attempting native bridge of SIP/192.168.1.7:5060 and
> SIP/213.137.73.140:5060
>
>
> Now: (Psychedelic phaser audio)
>       -- Called 666612192538552 at iconnect
>       -- SIP/iconnect answered SIP/ata1
>       -- Attempting native bridge of SIP/ata1 and SIP/iconnect
>       -- Attempting native bridge of SIP/ata1 and SIP/iconnect
>
> Hope somebody knows what I might try next. . .  I reverted to the old
> binaries and it works just fine again.
>
> Thx.
>
> B.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>