Archives : September-2015
Does anyone have any information for me?Welinghton.Citando Welinghton Magno Guimaraes : WELINGHTON MAGNO GUIMARãESDIRETORIA DE TECNOLOGIA DA INFORMAçãO – DIVISãO DE VOZFONE: (38) 3532-1285 OU (38) 3532-1200 – RAMAL: 8245 OU 8251UNIVERSIDADE FEDE..
I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years.The local telco finally got an ISDN switch and wants to move them to PRI.I didnt see this as a big problem – Ive done a few others on t..
How have the same sip username in several realms ?For now, I must add the realm prefix in the sip username of chan_sip.For example:[lg_2540]amaflags = default call-limit = 10host = dynamic language = en_UScontext = lg_default callerid = LG secret = XXXXXXXXXXXXXXXXXXXXXXXXXXt..
I am using the asterisk 13 and I config my dialplan for the SIP messaging as the following :http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html[astsms]exten => _.,1,NoOp(SMS receiving dialplan invoked)exten => _.,n,NoOp..
I have a VoIP provider that is requiring me to use an outboundproxy when connecting from Azure.I have set the outboundproxy set in the vendors section of my sip.conf file and when Asterisk attempts the registration, the attempt times out.I have mo..
I am using CentOS 6.7 64 bit and sox to convert gsm files to wav.When I do that I used to get a header like:RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 HzNow I get a header like:RIFF (little-endian) data, WAVE audio, GSM 6…
Ive built PJSIP a few months ago on a server that was 12.04 and cant remember how I got past this same issue. Ive looked at the links Ill put below and the comments section where others had the issue, but those tips arent helping either.Basically everyth..
Hi!I did some tests with Asterisk 11.19.0 and Confbridge. Ive wanting to migrate from Meetme to Confbridge for a long time now.For two participants in a conference – one actual call and one local channel that are recording – the cpu sits at 20%. ..
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I have and AMI application that tries to redirect a channel if a certain condition exists. It seemed to work when using Asterisk version 11.14, but now I am trying it with 11.19 and it is not. Here is the scenario: 1. A channel connects to the dialp..