Archives : February-2017
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Now that the g729 patents have expired, how do we use g729 in Asterisk?Will Digium be releasing a g729 codec for free use or do we download the free codec off the Internet now that we can use it without moral or legal res..
I noticed that when I dial some 7 followed by any digit, the other side gets confused. I would like to double the milliseconds inter-digits in SendDTMF(). Is there a way to change both the DTMF duration and its..
Trying to compile app_swift with Asterisk 14.2.1 and getting the following.Can anybody tell me what Im missing?: [root@localhost app_swift-master]# make ____ (_)/ __)_ _____ ________ ___ _ _ _ _ _| |__ _| |_ (____ |_ \|_ \ /___) | | | (_ __|_ _) / ..
Motty CruzProbablyit could become from missed configuration, check contexts issue. Check SIP context=sip-phone and extension dialplan context, probably you forget to set include or mistyping or other related to context issue.Mc GRA..
If I have a SIP endpoint defined in sip.conf using a host name instead of an IP address, do I have to reload sip to get Asterisk to re-resolve the host name if I change the IP address in my DNS?Does the answer change if the host name in sip.conf resol..
I operate an Asterisk server (v11.13.1) on Debian stable, and its rock-solid. The other day, however, I accidentally upgraded the kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped working.Below you can find my analysis while runn..
my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error:– SIP/voipeer-00000..
if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?in old SIP channel, we had ${HASH(SIP_CAUSE,)}but in PJSIP it has to be the outbound channel, which is gone when the control returns to the callin..
i have similar problem to https://issues.asterisk.org/jira/browse/ASTERISK-25806do you know about some workarounds/patches for better scalability?t..
I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, aftera call finishes the 2nd number get called and so forth.Im familiar with Asterisk/Elastix but the Campaign feat..