similar to: 1.8.4 quitting console

Hi! I've noticed 1.8.4 keeps quitting console by itself. Is this a bug or feature? :) Nick

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

All; I have a Fedora 26 server that I am trying to compile asterisk-certified-13.13-cert6 on. However, I'm getting the following errors. I'm also having a tough time trying to compile Dahdi. I'm not sure what I'm missing, but if anyone else is running Fedora, I'd really appreciate any help at all. Thanks Much; John V. make[1]: Leaving directory

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

I'm still not getting my outbound to work. I've seen two patches relevant to broadvoice for chan_sip.c which apparently have already been added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk doesn't seem to know they're gone though. I called my cell w/ broadvoice and turned on sip debug AFTer the call had physically dropped: *CLI> sip show registry

2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com> * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0 and under 1.6.2.16.1 and I get the same error: [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok [Aug 11 06:50:20] VERBOSE[3023] tcptls.c:?? == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 11 06:50:20] WARNING[3023] tcptls.c: FILE * open failed! Following the

Hi, I?m trying to get TLS to work with asterisk and client phones, and all I?m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to

Hey all, I'm currently running Asterisk 1.8.3 with FreePBX 2.8.1.3. It's tied to another IP-PBX via TLS. I have two problems going on.. 1.) Every so often (say roughly every 24 hours), Asterisk stops handing calls back to the second IP-PBX. The call rings indefinitely and Asterisk complains about the certificate like below: a. [Mar 16 16:10:04] VERBOSE[2973] tcptls.c: SSL certificate ok

The Asterisk Development Team would like to announce the release of Asterisk 13.22.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.22.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

The Asterisk Development Team would like to announce the release of Asterisk 15.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! I tried both signed and self-signed cert to no avail. Here is my Configuration: Sip.conf

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps?

Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Oct 26 14:38:19]

I've compiled guestfsd and run it on CentOS 6.3. It worked well, but when I quit guestfish, the guestfsd process in guest is always dead itself automatically. Is this a bug? or did I miss something? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://listman.redhat.com/archives/libguestfs/attachments/20120913/dac7ec76/attachment.htm>

The Asterisk Development Team would like to announce the release of Asterisk 16.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: