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The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

...=IN IP4 <my static ip> b=CT:384 t=0 0[Feb 10 12:04:20] WARNING[993]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 784523d570058f2f64315e506a79ee0f@<my static ip>:5060 for seqno 103 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [Feb 10 12:04:20] WARNING[993]: chan_sip.c:3415 retrans_pkt: Hanging up call 784523d570058f2f64315e506a79ee0f@<my static ip>:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). == Everyone is busy/congested at this time (1:0/0/1) -- Executing [08712...

...app_mixmonitor.c: == End MixMonitor Recording SIP/1002-00000292 [2013-09-25 13:58:31] WARNING[1380] chan_sip.c: Retransmission timeout reached on transmission 1aghl9fein for seqno 665 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response [2013-09-25 14:04:24] VERBOSE[32430] asterisk.c: -- Remote UNIX connection disconnected [2013-09-25 14:04:29] VERBOSE[1355] asterisk.c: -- Remote UNIX connection [2013-09-25 14:04:55] VERBOSE[1380] netsock2.c: == Using SIP RTP TOS bits 184 [2013-09-25 14:04:55] VERBOSE[13...