Dale Weaver

I have a SIP Trunk setup for SkypeConnect. Everything is working great. However, I can't get my CallerID to pass through remote phones I call. They SkypeConnect number is what is displayed. I have the Trunk ANI set but Skype doesn't seem to pass it on. Does anyone know how to get skype to pass along the caller id?

I'm on SnomOne 4.5. There is no drop down box to configure sip trunks. I didn't configure it per Nexvortex documentation.

I'm setting up a new SIP trunk with nexVortex.com. I can make and receive calls with them. However, I only ever have one way audio. Snom sends the audio out but I never receive any back. I called nexVortex and they think that the audio they are sending is going to my SnomOne's private ip address, 192.168.0.7. How do I change the SIP Packet to tell nexVortex to send audio back to our public IP address so the firewall can route it to SnomOne. Right now I don't have any of my public IP's being forwarded to SnomOne. With my previous SIP provider, this wasn't necessary. Here is part of the SIP trace they send me from their end. 2013/10/02 18:38:55.958292 173.167.75.17:5060 -> 66.23.129.253:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK7fe3.1c76 34.0..Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-fa339-524c6830-71d e681f-2ba94f90..Record-Route: <sip:17172834883@66.23.129.253:5060;nat=yes;f tag=c00080a-13c4-524c6830-71de681f-15d3a53;lr=on>..From: "WIRELESS CALLER" <sip:15514043819@208.94.159.10:5060;isup-oli=61>;tag=c00080a-13c4-524c6830- 71de681f-15d3a53..To: <sip:17172834883@66.23.129.253:5060>;tag=04a10ec64c.. Call-ID: CXC-87-7a3d5780-c00080a-13c4-524c6830-71de681f-662e588d@208.94.159 .10..CSeq: 1 INVITE..Contact: <sip:Ai11163mLaK@192.168.0.7:5060;transport=u dp>..Supported: 100rel, replaces, norefersub..Allow-Events: refer..Allow: I NVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE..Accept: application/sd p..User-Agent: snomONE/4.5.0.1090 Epsilon Geminids..Content-Type: applicati on/sdp..Content-Length: 274....v=0..o=- 20464 20464 IN IP4 192.168.0.7..s=- ..c=IN IP4 192.168.0.7..t=0 0..m=audio 60872 RTP/AVP 0 18 101..a=rtpmap:0 p cmu/8000..a=rtpmap:18 g729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephon e-event/8000..a=fmtp:101 0-16..a=rtcp-xr:rcvr-rtt=all voip-metrics..a=sendr ecv..

The call does transfer. That is not the problem. If I set it to immediate, the call will transfer immediately to the cell phone. What I want is for the office phone to stay ringing while the cell phone is ringing. It does this if I set the "Confirmation" field to "Press 1 to connect call". Both cell phone and office phone will ring at the same time. However, on the cell phone side we don't want to have to press 1 to accept the call. But when we set the "Confirmation" field to "No confirmation required" and not change anything else, the office phone stops ringing when the cell phone starts ringing. We want them to both stay ringing at the same time.

We have Snom setting "Include the cell phone in calls to extension" set to "After 5 seconds". If we have the "Confirmation" field set to "Press 1 to connect call", the desk phone will ring for 5 seconds, then both desk phone and cell phone start ringing. This is great. However, we do not want to press "1" on the cell phone to accept the call. We want to accept it as soon as we answer. Logically we set the "Confirmation" field to "No confirmation required". Now the desk phone rings for 5 seconds and then displays a message, "Call completed elsewhere" and stops ringing, then the cell phone starts ringing. This is not what we want. We want both phones to ring but without the cell phone having to press "1". That way if the cell phone doesn't answer, the voicemail is left on Snom. Anybody know how to fix this?