Archives : February-2014
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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..
Im using asterisk 1.8 as an answering machine. Id like to hear the calls it answers aloud in case I want to pick up and interrupt the call.There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I coul..
We have a customer reporting poor quality calls when they come to us via a particular provider. The SIP traces look perfectly normal both on the ingress to us and egress to another telco. No additional sip messages after the call has been answered un..
Dear Folks,Im not able hearing the voice of client but on other hand client able to hearing my voice.Im not able to find out the problem where is im wrong.Im getting continues following error:chan_sip.c:10391 check_via: is not a valid hostConfigurat..
Which is the best way around to integrate Asterisk with VoiceXML like VoiceGlue…! Am using Asterisk 11.2…