Archives : October-2014
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, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network.i have done settings as belowres_stun_monitor.confsettings:[general]stunaddr = stun.ideasip…
When I issue the CLI command core show calls I see how many calls have been processed by Asterisk since it started; eg:0 active calls198 calls processedIs there a way to reset the calls processed counter without having to shutdown and restart..
hi.i have dialplan with 2 simultaneous calls – dial(sip/phone1&sip/phone2).when i cancel call on phone1 (push reject button), the call is still ringing on phone2can i cancel call on both phones from one place(one pho..
I have Asterisk 11 with DAHDI (Sangoma E1-Card) running on Ubuntu 12.04LTS. Asterisk and DAHDI-Drivers are installed from source.When doing an apt-get upgrade the system packages will be update but sometimes Asterisk is broken. Which packages do I h..
It might be the case that you are are trying to use SIP client over 3G and It registers and call can be initiated from the client but it cant receive call; cause asterisk sever marks it as unreachable immediately after registration. Even more, the ab..
I have added the following to the peer definition :ignorecryptolifetime=yesBut still Asterisk tells me :[Oct9 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[O..
Does anyone know how frequent segment faults occur in the current LTS release (version 11) and in the future LTS release (version 13)?We are currently using 1.6, which frequently throws unexplained segment faults, thats why we are considering to upgr..
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+TutorialBut when I try to make a c..
——=_NextPart_001_0057_01CFE227.DD49D8C0Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: 7bitDear Sir/ Madam, My client is using Asterisk based telephony system. He has 300 concurrent users with two PRI Channels. He is request..
Darren has recently merged my changes to app_swift – now supporting asterisk 12 and 13.If anyone has the Cepstral TTS engine installed and would like to link it with asterisk, app_swift is the way to go.This is the first version that configures to m..