similar to: Asterisk SSL support broken with update from openssl-1.0.0 to 1.0.1e, recompiling does *not* help

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure:

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has

Hello, I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. I followed this: cd /usr/src wget ... asterisk-13.13.1.tar.gz tar zxf asterisk-13.13.1.tar.gz cd asterisk-13.13.1 ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr" ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled make menuselect (shows res-srtp is available) make latest make command fails with

hi all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

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

The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.16.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.5.0 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:

Hi, I have been running TFTPD server for 3 weeks and performed about 100 deployments. After that, TFTPD started throwing the follwing errotrs in /var/log/messages Aug 20 21:52:55 RTP-OSP-Server tftpd[7146]: warning: cannot open /etc/hosts.allow: Too many open files Aug 20 21:52:55 RTP-OSP-Server tftpd[7146]: warning: cannot open /etc/hosts.deny: Too many open files Aug 20 21:52:55 RTP-OSP-Server

Hi, my log is full of errors from this mobile user: -- Registered SIP '0010106' at 212.93.97.135:7759 [2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804 handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms / 10000ms) [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...): Name

Patch it by hand. Follow this help http://www.asteriskguru.com/tutorials/spandsp.html Hi Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Tomislav Parcina Inviato: venerd? 13 gennaio 2006 12.28 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] RE: RE: Spandsp In article

*************** *** 41,50 **** APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so a pp_zapbarge.so app_zapscan.so" ; fi) APPS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so a pp_zapbarge.so app_zapscan.so" ; fi) APPS+=$(shell if [ -f /usr/include/osp/osp.h ]; then

So I've been trying to use this site and the daemon script here. http://www.linuxjava.net/howto/webapp/#tomcat It works so far. I mean, at least the server starts, I can get to the site, etc. So far, so good. The only problem is, for some reason when I start tomcat using root everything runs fine. When I start tomcat using this script, however, I get this.... FATAL [http-8080-Processor25]

I am currently running Asterisk 13.1.0-1 I have a chan_sip configuration that works fine with a 3rd party. Third party does not use authentication or registration, it's ip based authentication... When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk side. What has me really baffled is the debugging indicates [Jul 14 17:28:24] DEBUG[3620] pjsip: sip_endpoint.c

Hi there! My issue is that i have (bouht to me) a box with Asterisk 1.8.2.2 but its seems that with no SRTP support. So i added a libsrtp libraries and like i understand now i need to recompile/reinstall Asterisk... is that safe to all data and MySQL backand data? -- Best regards Matiss Jekabsons Procerto Ltd. Dzelzavas Str. 117. Riga, Latvia -------------- next part -------------- An HTML

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip:

Hello! We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have several other customers running both versions. The customer in question does not use us as their provider as they?re located in a different country. When they make outgoing calls, there is a 3 second delay between answering the call and the call being established. When debugging this, I found that Asterisk

Hi Timo, I'm curious and have questions about the new Object Storage Plugin (OSP), and how it can be leveraged by an SMB like us. First, am I reading this right where it could be used as a kind of 'live/realtime backup' solution, where everything is stored *both* locally and in the cloud, with two-way syncing, ie, so local users could access the local server for faster access,