Ive been getting slammed with these messages on my console lately.ed -1: Invalid argument[2016-05-31 10:09:40] WARNING: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0 returned -1: Invalid argument[2016-05..
Over the past week weve noticed that, when bridging a local channel created with ARI, audio processing uses excessive amounts of CPU time. After some digging, we determined the cause, have some recommendations, and have a patch up for review.The cause:W..
Calling linphone from asterisk 13.9.1.:Dial(SIPfirstname.lastname@example.org)And it works. But on the linphone side the caller is:@email@example.comIs there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALL..
everyone.It seems that all the documentation for Asterisk has become obsolete when it comes to using the Monitor command on a call queue.To the best of my knowledge, the way to get Asterisk to record a call that goes into one of your call queues is..
Asterisk 13.9.1 seems to be ignoring my realtime IAX configuration.I have carried this configuration over from version 1.8 and it worked until 13.7 at least.The config mapping is done:pbxoficina*CLI> core show config mappings Config Engine: mysql=..
folks,At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow th..
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to everybodymy system is be attack, but I dont know what this means[May 27 15:12:24] WARNING chan_skinny.c: Partial data received, waiting (76 bytes read of 786)[chan_skinny.c] skinny_session[C-00000000] skinny_session:WARNING[May 27 15:52:..
I have two Polycom phone configured with Asterisk server, both use transport=tls My provision server is FTPS, I have phone5006.cfg & phone5007.cfgIf I enable transport tls on both phones I get the following error:[2016-14:03:52] WARNING: chan_sip.c:14..
Original Subject: Questions… connecting Asterisk to the WorldIn short, adding the line > overlapdial=yes in chan_dahdi.conf changed everything!!I wrote:Without..