On 03/15/2011 10:18 AM, Paul Belanger wrote:
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
[email protected]:5060 for seqno 102
(Critical Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
Well... everything works fine, but I don't like this errors, any ideas?
Theses are leftover issue with the IPv6 conversion for Asterisk 1.8.
Collect a complete debug log[1] and open a new issue on the tracker.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Umm... no, I don't think they are. His log showed an attempt to use
Dial() to connect to 'SIP/h', and he had no peer named 'h' or is that an
IP address or DNS name. It should have failed a little more cleanly than
it did, but I'm sure that at least part of the problem is attempting to
dial a SIP endpoint that doesn't exist (and dialing out from the 'h'
extension as well).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
