similar to: sdp_crypto_process: Crypto life time unsupported: crypto

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

Hi all, i try to deactivate SRTP in asterisk 11. In sip.conf: tlsenable=no encryption=no transport=udp srtpcapable=no but when I try to make a call comes following message: [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1

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

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

The Asterisk Development Team has announced the release of Asterisk 1.8.29.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.29.0 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 1.8.29.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.29.0 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 11.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.11.0 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 11.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.11.0 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 folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce,

Help solving a performance mystery would be welcome. I have two test programs: * one totally CPU bound (it just infinitely loops) * the other I/O bound (it loops, each time doing * a write, * an fsync or fdatasync (I get similar behavior with either), * an lseek to back up to the same position again Under ext2, if I run these two together, they each run nearly as fast as

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

I have a column in a data frame that has a class of "Date" and a mode of "numeric". When I: max(df$Date) My output stays in Date format, i.e. "2006-01-03". However, when I run the following statment: tapply(df$Date, df$SomeFactor, max) my output looks like this: 9129 9493 9861 10226 10591 10956 11320 11687 12052 12417 The returned object is of

Hi there, need some helb please: I build a setup with two samba4 AD servers. Unfortunately there is a cadaver and doubled servername in the site-config and I am not able to delete it: root at sambak26:~# ldbsearch -H /var/lib/samba/private/sam.ldb '(objectGUID=cf7d8ac1-b0ae-4e72-9129-ed480ee38006)' --cross-ncs -d0 # record 1 dn: CN=NTDS