search for: voicetrad

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=...

On Thursday 18 June 2020 at 19:57:03, Administrator wrote: > does some people here use https://voicetrading.com which is a Dellmont > service from Netherlands. At the high begining they were also known as > Finarea (CH and DE mixed Co) > Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or > equal to CALLERID(num). We tried replacing + with 00, same problem. > > Ther...

Hi, sorry for this a bit OT. I'm using VoiceTrading for some calls -premium route- and can't get CID to work despite the fact that CALLERID(num) and CALLERID(name) are setted. I ask in VT->myAccount to accept calls from my IP without checking username & secret. On incoming calls the CID is setted to 0100000000 If I accept calls fro...

...really picked up the phone from the other side! The message is played randomly and appears once every 4-5 call attempts. You can get it with the first call or if you call 4-5 times the same number. The customer learned about this scam about a month ago using a wholesale termination service called VoiceTrading: http://www.voicetrading.com. Thank you Ben for reporting this to us!" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/c5bd841f/attachment.htm

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider to within USA. So, we bought the minutes from Gafachi. Still we are having the call latency issues. $ConversationFile = $ConversationPa...

...ent for ID 101 Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 62.xx.xx.xx:8786 -- SIP/Voicetrading-08e1ce18 is making progress passing it to Zap/5-1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080416/c74bb16c/attachment.htm