Archives : February-2014
Home » 2014 (paged 48)
Im using py-Asterisk 0.5.3. Im trying to use it along a Tkinter-based GUI so Ive dedicated a thread for reading incoming AMI events.Which is the preferred way to gracefully exit from an event loop ?More precisely:This thread is waiting for input eve..
Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in make menuselect so I dont understand why its showing as no translation path. Any ideas?I am running certified-asterisk-11.2-cert2Tha..
Dear Folks,[Test_Context]exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log)exten => _911.,2,Set(CALLERID(num)=xxxxxxx)exten =>_911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})ex..
Were running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now.But!Weve got such a incoming call volume over the few weeks that well have Asterisk occasionally restart itself.My hunch is that it is in part memory pressur..
Im configuring a brand new polycom SSIP 7000.To my surprise, when this telephone boots up, my DHCP server receives a request that Wireshark classifies a BootP request from which I cant find any Vendor identification.The trouble is my DHCP server u..
2014-02-06 11:09 GMT+01:00 giovanni.v :Adding priindication = outofband in appropriate file gave what I was looking for.Thank you very much for shar..
Im looking for some beta testers to provide feedback on an Asterisk intrusion detection & prevention program were releasing soon.As a quick overview, the program provides:- banning based on geographic location of source IP (Continent, country, regi..
I am using voip with Vodafone as SIP peer for outbound telephony and i have a huge problem establishing calls to other people. It works like in 1 of 5 tries. The peer is sending SIP 480 temporarily not available. It took a while to identify this, beca..
Im trying to address a problem with users transferring to invalid destinations.In my sip peer Im setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred.[customer-forward]ex..
All;Im running Asterisk 1.8.15-cert3 with the newest version of spandsp. Ive even tried unloading that and using Digiums FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_ty..