search for: voip3

...plete the call transfer. This frees up the analog phone line to process another call and the central office handles the transfered call. This transfer feature can be done with a channelized winkstart T1, and is possible on a PRI. On a PRI, this feature is called a 2B Channel Transfer. Contact us at voip3@nibble.net.

...got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format slin since our native fo rmat has changed to gsm I have set ALLOW=ALL in iax.conf (on both asterisk servers) , sip.conf on the local server. I have tried setting allow=ulaw (Only) and get: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format s...

...,SetVar(ARG3=${ARG4}) exten => s,7,SetVar(ARG4=${ARG5}) exten => s,8,SetVar(ARG5=${ARG6}) exten => s,9,GotoIf($["${GATEWAY}" = "voip1"]?14) exten => s,10,GotoIf($["${GATEWAY}" = "voip2"]?18) exten => s,11,GotoIf($["${GATEWAY}" = "voip3"]?16) exten => s,12,Macro(dialout,SIP/${ARG1}@pstn) exten => s,13,GotoIf($["${ARG3}" = ""]?20:5) exten => s,14,Macro(dialout,IAX2/voip1/${ARG1}) exten => s,15,GotoIf($["${ARG3}" = ""]?20:5) exten => s,16,Macro(dialout,IAX2/voip2/${ARG1}) e...

...hine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend host=10.0.0.160 auth=rsa secret= inkey=voip3 outkey=voip1 context=CONTEXT_VOIP1 allow=all ipaddr=10.0.0.160 port=4569 qualify=yes trunk=yes The iax.renoir.conf file contains : [VOIP_RENOIR] type=friend host=renoir.lucyde auth=rsa inkey=key_184 outkey=key_Renoir context=CONTEXT_RENOIR trunk=yes allow=gsm Thanks to the variable context, whe...

We need an expert in least cost routing (LCR) for an Asterisk project. Please provide references and a resume of your experience. Contact us at voip3@nibble.net.