Displaying 5 results from an estimated 5 matches for "dumphistori".
Did you mean:
dumphistory
2007 May 03
2
SIP peer / Maximum retries exceeded on transmission
Hi Everyone,
I was hoping someone might know why I am experiencing a problem with
Asterisk logging the event:
[May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response)
This is happening after:
- call is setup, 2 way audio
- call can function correctly for up to 5
2010 Nov 03
1
inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
<--- SIP read from UDP:147.135.32.221:5060 --->
INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0
Call-ID: 31007e-31 at 147.135.32.221
CSeq: 1 INVITE
From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
To: "Gregory Malsack"<sip:s at
2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
Hello
Have a setup of asterisk with realtime SIP devices.
Trying to organise monitoring of my SIP devices. Once device
registered, its state becomes NOT_INUSE (result of
DEVICE_STATE(SIP/device) function).
Simulating of device breakage - powerdown it.
Waiting for a while (minute or two), retrieving
DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE.
doing from CLI:
sip qualify peer
2010 Oct 12
0
rtpip patch
Hello *,
is the rtpip patch still valid for asterisk 1.6 (with some code
changes, obviously)?
https://issues.asterisk.org/view.php?id=8161
Or, in asterisk 1.6 there is an alternative to using it?
This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11
--- chan_sip.c 2010-10-12 13:47:49.000000000 +0200
+++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200
@@ -987,9 +987,6 @@
2010 Nov 05
2
How to append custom option to Contact: header on outgoing SIP INVITE msgs?
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here>
where SITE-ID=us.here is set in a config file that * parses on
startup. Or in a Dial() command option? Or I don't care exactly
how. :-)
It is possible to