Displaying 20 results from an estimated 3000 matches similar to: "Repeated Locally bridging messages"
2013 Sep 17
1
RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.
Here is the sip peer information for the call
2013 Sep 27
2
Is this SDP payload Asterisk created valid?
We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network.
After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488
2016 Feb 10
2
Unexpected termination of the call when pick up (res_pjsip)
Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk
suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web:
http://pastebin.com/tLNCpx4d
No diagnostic messages why asterisk suddenly decided to hangup i don't
found :(
There are suggestions or strong belief
2015 Sep 08
2
Network range in trunk definition
I have some problem finding a smart way to add inbound trunks ip
authentication. I don't want to set allowguests=yes
Some of my providers just list some IP and I add them like:
[provider](!)
context=fromoutside
type=friend
insecure=port,invite
disallow=all
allow=g729
allow=ulaw
allow=alaw
canreinvite=no
[magrathea1](provider)
host=87.238.72.129
[magrathea2](provider)
host=87.238.72.130
2012 May 09
1
No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Hi,
I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!...
This is the SDP portion that comes in the INVITE messages of calls
2013 Aug 27
1
Asterisk 1.8.15-cert3, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.5.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones,
and 11.5.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of
2017 Feb 02
0
Dovecot performance and proxy loops with IPv6
Hello list,
i run here an large mailsetup with some million mailboxes and got strange performance problems, cause i think i have overseen or forgotten an simple setting.
Here are some details:
21 CentOS 7 Servers with dovecot 2.2.25 and ldap userdb/passdb via socket behind an hardware loadbalancer.
The storage behind is an ISCSI Storage with 4 10Gbit/s multipath paths, splitted up to 10 TB
2013 Aug 27
0
Asterisk 1.8.15-cert3, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.5.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones,
and 11.5.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of
2016 Aug 15
0
Certified Asterisk 13.8-cert2 Now Available
The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 13.8-cert2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the
2020 Apr 30
0
Certified Asterisk 16.8-cert2 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert2.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2020 Apr 30
0
Certified Asterisk 16.8-cert2 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert2.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2017 Feb 08
0
define auth timeout
Hello list,
when reloading dovecot via "doveadm reload" it throws me tons of errors like this:
Feb 08 10:57:30 server1 dovecot[18243]: Feb 08 10:57:30 imap: Error: net_connect_unix(/run/dovecot/auth-master) failed: Resource temporarily unavailable
Feb 08 10:57:30 server1 dovecot[18243]: Feb 08 10:57:30 imap: Error: net_connect_unix(/run/dovecot/auth-master) failed: Resource temporarily
2015 May 21
1
asterisk 13 webrtc
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in
SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
2014 Feb 11
0
g726 transcoding
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
I am running certified-asterisk-11.2-cert2
Thanks
Gareth
> core show translation paths alaw
--- Translation paths SRC Codec "alaw"
2013 Mar 27
0
Asterisk 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, 11.2.2 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of these versions
2014 Oct 20
0
Asterisk 1.8.28-cert2, 1.8.31.1, 11.6-cert7, 11.13.1, 12.6.1, 13.0.0-beta3 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1,
11.13.1, 12.6.1, and 13.0.0-beta3.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of
2015 Apr 08
0
Asterisk 1.8.28-cert5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, 13.3.2 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The
2017 Dec 22
0
Asterisk 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2 Now Available (Security)
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
2018 Jun 11
0
Asterisk 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2 Now Available (Security)
The Asterisk Development Team would like to announce security releases for
Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
13.21-cert2.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
2013 Mar 27
0
Asterisk 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, 11.2.2 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of these versions