Archives : October-2014
I am a newer on asterisk. when I tried to send fax, I can not get success. After doing a lot of reseach, I decide to ask my question at here.Instead of using old fax machine, I want my system can send fax via t.38. Could anyonegive me some idea to corr..
HiWill the presentations made at Astricom 2014 be made public as recorded videos ?tha..
is it possible to reload just a context in stead of the whole dialplan ?I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934But is it possible in some Asterisk version ?Kind rega..
HiThe Asterisk 13 is already stable for production environment?thanks[image: Sua Foto] Rafael S. SaraivaPorto Alegre – RS| Mobile:(51)..
I am new to Asterisk forum :).I have a requirement of terminating3G Mobile originated calls (coming through 3G-MSC)to EPBX Phones via Asterisk PBX.Setup:Mobile—-> Mobile Switching Center ( 3G)—–SIP interface—>Asterisk PBX—>SIP Phone.I wan..
Asterisk users, We noticed that on Asterisk 12 zombie processes are being generated – They are released after a while, but we have around 10-20 zombie processes running.We are not sure if this is a normal behavior or an issue.We saw in the documentat..
Howdy,Im trying to get my feet wet with pjsip using the conversion script mentioned on the Wiki on this page:https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsipIm using the copy of the script thats included with Asterisk 13/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjs..
I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonferenceThat said Asteris..
all,what should i do if i want to know how long asterisk server take a time for registration 1 client on server side?especially just for voip server authentication, when we have to registered username and password in sip.conf and extensions.conf fil..
Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached?..