similar to: deactivate SRTP in asterisk 11

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

Hello, I have added the following to the peer definition : ignorecryptolifetime=yes But still Asterisk tells me : [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32 [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process: SRTP crypto

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]

Hi all, is it possible that asterisk uses two proxies with SRV? The enddevices are registered on one of the two Proxies (Kamailio). The two proxies communicate with each other. And asterisk can choose one of this proxies with SRV. asterisk | \ | \ Proxy1 Proxy2 I have tries to solve this problem with two trunks for this proxies and Dial(... at proxytrunk) but on this way the

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=-

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm running Asterisk 11.3.0 on wheezy. I'm trying to do TLS +SRTP with blink SIP clients as shown here https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. TLS is fine and I can call between clients. SRTP is a different matter, my SIP clients return: SIP 488 "Not acceptable Here" I'm really stumped on this

Hey there i'm trying to get an Asterisk 11.11 with encryption working with my Grandstream phones. But i stuck. To avoid NAT problems i'm using IPv6 Just with TCP/TLS it's working fine. Only the SRTP funktion is not working. Asterisk tells me WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fa10800f5a0 (len 681) to [2a02:1205::...]:37635 returned -2: Success and also SSL

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

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

How do you make a button inactive, so that it is visible, but it will not respond to clicks? Thanks Jayson

Hi, I just created a new Rails app that will be receiving some POSTed data from the outside so it must skip the verify_authenticity_token for some create actions. Although I have added: skip_before_filter :verifiy_authenticity_token I still get InvalidAuthenticityToken. In one of my other Rails app (created back in Rails 1.2.6 and updated to 2.3.2 over time) this skipping works perfectly though,

virsh provides net-start <netname-or-id> to activate an inactive network. Does virsh provide any way to do the exact converse? deactivate an active network without destroying, undefining, removing, ... Also - I am curious - what does activate actually do? What change in state of anything occurs? ( I read the document at http://libvirt.org/formatnetwork.html which

Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before

Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus

Hi, we have following setup: fax machines > PSTN -> GW SIP -> Asterisk -> Peer IAX (Elastix) Asterisk is 11.16 as well as Asterisk version of Elastix peer. When sending incoming fax calls to the Peer IAX -which receive them using hylafax- we want to tell our asterisk to NOT detect fax CNG. It's easy on a SIP channel as faxdetect=no can be set for the peer, but how to do it