similar to: Asterisk 11.8.0 and 11.8.1

Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip reload" or "iax2 reload" does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

Hello, I have tried to install openssh 2.9p2 on Solaris 2.6 and Solaris 2.8 after compiling the openssl (both 0.9.6a and 0.9.6b) and openssh software with both gcc 2.95.3, Sun's Forte compiler, Sun's linker, etc. I have tried the zlib from Sun freeware and version 11.8.0 REV=2000.01.08.18.12 contained in the SUNWzlib package. Both openssh and openssl appear to make properly and all the

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent

We're experiencing an issue where calls disconnect after 15 minutes. It seems to happen just after Asterisk sends an update mesage. RTP is being set up directly. Asterisk is only in the SIP dialog. Has anyone experienced this issue? 4 PRIs inbound, 4 PRIs outbound, asterisk provides switching. SIP/2.0 200 OK Via: SIP/2.0/UDP 38.XXX.XXX.XXX:5060;branch=z9hG4bK1c4b524f From:

Le 08/07/2015 17:36, Richard Mudgett a ?crit : > > > On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hi list, > > we wanted to patch our servers with 11.18.0 patch against 11.17.0 > actual running version. Patch failed with > > zone-s:/usr/src/asterisk-11.18.0# patch

For a future release of Portable OpenSSH, it would be nice to have a ./configure option to enable the binaries produced, to be statically linked. I tried using LDFLAGS option to ./configure .... but this passes arguments to gcc not ld, this should be documented. Additionally, I feel that ./configure should --extra-inc=dir and --extra-lib=dir to add paths to compile (-I) and link lines (-L &

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading "User Profile Configuration Options" the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly.

Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick

Hi Patrick, Thanks a lot for your quick help. Yes, I configured the NAT options in sip.conf. BTW, I am using 12.1.1, will try 11.8.1 and see if I can make it work. Thanks again, William ======================================= Date: Sat, 05 Apr 2014 23:38:32 +0200 From: Patrick Laimbock <patrick at laimbock.com> To: asterisk-users at lists.digium.com Subject: Re:

Hello, I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the same result. What I did: dial is done like exten => s,n,Dial(SIP/<IP gw>/<dialed number>,,M(myMacro)), which tell Asterisk to

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

I am having an issue that prevents WebSockets over SSL/TLS (or any kind of encrypted HTTP traffic to Asterisk) from working after an openssl library update. My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP port 8089 initializes correctly with asterisk-11.7.0. After an upgrade to