Archives : February-2014
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..
all,I have an SPA112 that in sitting behind a Ubee cable modem.The internet link is solid, but the device becomes unreachable within a day or so of being rebooted.Then the customer goes to reboot the device, they report that all 4 lights are lit…
I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. Id like to use the multicastRTP driver to do paging.However, when a phone dials an extens..
– I figured this was probably the best place to ask this questionIs there anyone that has done anything practical with the API and/or Real Time Database config?If so, I would like to pick your brains if I may…
Asterisk,I just got V12 running and all seems well but just now I looked at my CDR logs and they were messed up so I copied over the sample cdr_custom.conf and uncommented the first master line and the simple line and the logs look like:Simple.csv:1391652220,,Master.csv,,,,,,,,,,,,,,,,,,..
,I have a provider that uses 6060# as a prefix, but when I send the INVITEasterisk is changing the number to 6060%23.I have activated pedantic=yes in the sip.conf but it seems not working at all.I have asterisk 11.7.0.Can someone please guide me here?Thanks,Wil..
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the${CDR(start)} is not returning any data. Other functions, like${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning correct values. Where is my mistake? Has t..