Brian Capouch
2003-Feb-15 03:16 UTC
[Asterisk-Users] Latest CVS freakout on iconnect calls
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.
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 >