d1rage5

I think this is a issue with axon itself, because axon does the same thing with ip voip providers when the call is transfered nothing is heard on the callers end but dead air. I think NCH already knows and has plans to correct in some future version... But who knows, i just wanted to give you a reply. i hate the no replys. Maybe if NCH are reviewing they can speak to this....

I will try it give it a shot,but its been a year sense i set this up. What type system is this running under? xp? 2k or 2k3 server? you say when receiving a call uplink says incoming call....... this would mean that a call is coming in through SKPYE? Right? and that the uplink see the call coming in but then it goes no where basicly? If this is true - check AXON - external line for Skype and see what it is set to ring on? How are trying to dial out with skpye? A SIP phone - what do you dial to access the skype line? Do you have speed dial numbers setup in skye? Like mine i have the prefix set to 6 in the dialing plan, so you dial 6 then the speed dial number and you can dial users skype name, if you have setup to dial land line (POTS Lines) then you would need a balance in skye or it will not allow you to dial out except to users that have an account. Lets start from here and if you have issues - you can PM me to see if i can take a look see for the issue.

on the server with axon make sure you have skype installed and auto login. install uplink skype will pop up that another program is trying to access it (accept this connect always from the uplink) Now in Axon: goto to creat an external line: the uplink should be added automatically, you just need to edit a few things according to your setup. (Incoming Calls Ring on Extension or Group) if you only have one line for the IVM then the default setup would be 198 to pick up calls coming in. to dial out using skpye: again in axon goto dialing plan and edit the default setup (here uplink should already be added) if not click ADD Dial Rule. (If number starts with (6) (or what ever number you want to use, as long as its not one already in use)). REMOVE DIGITS (1). PREPEND (+). DIAL ON LINE (what ever you names it on the external line (DEFAULT IS(Uplink Sip To Skype)) now on your sip phone just dial 6 then the number. To Dial a skype name and not a number (this takes some intervention) in skype you will need to setup speed dial numbers for each person that in the skype) then you would dial 6 and the speed dial number. i think this is about it its been a year now sense i set this all up.

iam useing callcentric but (yes the big BUT) they changed a little and not for the better on outbound calls. you get three trunks (meaning three calls at one line with one number) if you make more then one outbound call at the same time the second or third outbound call will be charged at the current per minute rate and not the unlimited plan you purchase. But they have great quality so far. they do support VOIP 911, ad you will be charged for this service if you want it or not. hope this helps

The text file can be created with notepad like so (100,101,102,103,104,105,106,107,108,109,110,111,112) THis would be all your active extions used just use the numbers and not the ( ) save the file with what ever name you want in IVM edit the OGM for each external line not the extensions. click the top tab (Key Response) click the box for (Limit acceptable values using list) Inclusive List Comma Delimited text file (browse to the file you create) click ok and thats it. when someone dials a line not used it will say thats not a valid ext...

I had this working at one time but it was hit or miss and then not all fax mahines support this very well. I finally went with an outside source that gets you a local or 800 number and faxes come to you by email or you can login and pull down the stored faxes..cost anywhere from 5 to 40 a month its according to the service provider you get.. I used faxmicro.com and got an 800 # 10$ a month flat fee no per page or per minute charges (local number was only 5$ a month). NICE thing is NO BUSY signel so your can receive a lot of faxes at the same time. Hope this helps.

Is there a way in the sotware to do bandwidth throtting? right axon uses 64kb per packet per cann ans stardard VOID uses 20 to 32 KB percall?

log from axon around this call 13:51:49 UDP Packet Sent to 66.168.184.122:5070 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.168.184.122:5070;rport;branch=z9hG4bK92856 To: <sip:105@97.81.16.235>;tag=4748 From: "spare 1" <sip:110@97.81.16.235>;tag=3612 Call-ID: 1216489815-2856-SPARE1@66.168.184.122 CSeq: 111 INVITE User-Agent: NCH Swift Sound Axon Virtual PBX 2.00 Content-Length: 0 ---------------------------------------------------------------- 13:51:50 UDP Packet Received from 10.33.15.214:5060 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 200 OK To: <sip:105@term1.pcspecs.net>;tag=3266fcc963117a45i0 From: "spare 1" <sip:110@97.81.16.235>;tag=4746 Call-ID: 1215625117-1712-TERM1@97.81.16.235 CSeq: 626 INVITE Via: SIP/2.0/UDP 10.33.15.45:5060;branch=z9hG4bK332551712 Contact: "Daniel Murray" <sip:105@10.33.15.214:5060> Server: Linksys/SPA942-5.2.8 Content-Length: 210 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 57460219 57460219 IN IP4 10.33.15.214 s=- c=IN IP4 10.33.15.214 t=0 0 m=audio 16418 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ---------------------------------------------------------------- 13:51:50 UDP Packet Sent to 66.168.184.122:5070 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.168.184.122:5070;rport;branch=z9hG4bK92856 To: <sip:105@97.81.16.235>;tag=4748 From: "spare 1" <sip:110@97.81.16.235>;tag=3612 Call-ID: 1216489815-2856-SPARE1@66.168.184.122 CSeq: 111 INVITE User-Agent: NCH Swift Sound Axon Virtual PBX 2.00 Contact: <sip:105@97.81.16.235:5060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Accept: application/sdp Supported: replaces Content-Type: application/sdp Content-Length: 280 v=0 o=- 57460219 57460219 IN IP4 10.33.15.214 s=- c=IN IP4 97.81.16.235 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=direction:active a=domain:97.81.16.235 a=local:10.33.15.214 16418 ---------------------------------------------------------------- 13:51:50 UDP Packet Received from 66.168.184.122:5070 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< ACK sip:105@97.81.16.235:5060 SIP/2.0 Via: SIP/2.0/UDP 66.168.184.122:5070;rport;branch=z9hG4bK102856 To: <sip:105@97.81.16.235>;tag=4748 From: "spare 1" <sip:110@97.81.16.235>;tag=3612 Call-ID: 1216489815-2856-SPARE1@66.168.184.122 CSeq: 111 ACK Max-Forwards: 20 User-Agent: Express Talk 2.02 Proxy-Authorization: Digest username="110",realm="axon@term1",nonce="v17109qaq88735w",uri="sip:105@97.81.16.235",response="baefee37ce0c187a56cc5fde0f4dc01c",opaque="" Content-Length: 0 ---------------------------------------------------------------- 13:51:50 UDP Packet Sent to 10.33.15.214:5060 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK sip:105@10.33.15.214:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.45:5060;rport;branch=z9hG4bK332561712 To: <sip:105@term1.pcspecs.net>;tag=3266fcc963117a45i0 From: "spare 1" <sip:110@97.81.16.235>;tag=4746 Call-ID: 1215625117-1712-TERM1@97.81.16.235 CSeq: 626 ACK Max-Forwards: 20 User-Agent: NCH Swift Sound Axon Virtual PBX 2.00 Content-Length: 0 ---------------------------------------------------------------- 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:50 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:53 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:56 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:56 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:56 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:56 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:57 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:57 Rtp Relay 66.168.184.122:8002 >>> 10.33.15.214:16418 (172 bytes) 13:51:57 UDP Packet Received from 10.33.15.214:5060 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< INVITE sip:105@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.214:5060;branch=z9hG4bK-c40b78eb From: <sip:105@term1.pcspecs.net>;tag=3266fcc963117a45i0 To: "spare 1" <sip:110@97.81.16.235>;tag=4746 Call-ID: 1215625117-1712-TERM1@97.81.16.235 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Daniel Murray" <sip:105@10.33.15.214:5060> Expires: 30 User-Agent: Linksys/SPA942-5.2.8 Content-Length: 229 Content-Type: application/sdp v=0 o=- 57460219 57460220 IN IP4 10.33.15.214 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 16418 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendonly ---------------------------------------------------------------- 13:51:57 UDP Packet Sent to 66.168.184.122:5070 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> INVITE sip:110@66.168.184.122:5070 SIP/2.0 Via: SIP/2.0/UDP 97.81.16.235:5060;rport;branch=z9hG4bK332571712 To: "spare 1" <sip:110@97.81.16.235>;tag=3612 From: <sip:105@97.81.16.235>;tag=4748 Call-ID: 1216489815-2856-SPARE1@66.168.184.122 CSeq: 626 INVITE Max-Forwards: 20 User-Agent: NCH Swift Sound Axon Virtual PBX 2.00 Contact: <sip:105@97.81.16.235:5060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=- 57460219 57460220 IN IP4 10.33.15.214 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 16418 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendonly

we have internal phones that call each other all day long no problem. when i sales person calls in from their voip phone (external to network) to and voip phone (internal network) the phone rings and is picked up you can hear the out person talking but they can not hear the internal person,,,,hears the catch if you hlace the caller on hold and take them off hold then you can hear both parties. using linksys SPA942 voip phones. also MOH internal it works great when a person calls in from a non voip line to our system the MOH works but that same (or any external voip line to the system) MOH does not seem to work?

do you know what plug in would be best for this?

IVM when you have a phone setup as it stands right now, anyone can walk up to anyones desk phone dial 199 and listen to that persons voice mails and also delete them. setup of password when phone is setup for voicemail. allow setup of password (numeric only) through the first connect to the voice mail. not to forget to have an admin way to reset this password as needed but to to see what is set.

the external line (xxxxxxxxx) comes in it will then routes to IVM (extension 198) the caller hears the announcement and make the choice of who they want to call (employee number or department) I have some departments that when they dial that department it needs to ring a set number of lines not just one desk phone. Being that there is only one external line (with multi trunks) the caller has to be routed to the ivm attendant so they can transfer to an extension or department. So yes if a caller dials in and ivm transfers them to an extension (employee or department) can that extension then dial (multi extensions or a group of extensions)?

I dont think anyone is getting what i am wanting.. 1. When a customer is calling a an external line (XXXXXXXXX) this line comes into axon will ring on group (701) and ext (198) the ivm auto attentment picks up and will route the call to the imput extion the called pushes. 2. When caller pushes ext (XXX) they will be routed to that extion. Can that extion dial a group after being routed to the extion? Aging i do not want the main call to RING on a group of numbers just the extion ring a group after being routed to that ext. Does sound better to what i am trying to ask?

Where is this setup? i know you can create a group or queue but when someone dial ext 105 how do you set it to dial that group and still leave boice mail in 105's voice mail if no answer. i dont want the exteral extline to dial in to a group.

Where is this setup? i know you can create a group or queue but when someone dial ext 105 how do you set it to dial that group and still leave boice mail in 105's voice mail if no answer. i dont want the exteral extline to dial in to a group.

Like having a group line dial multi lines at one time can you do the same thing with an extion number? someone calls in 198 picks up the call they transfer to an extion number 105. can you make 105 ring a group of internal lines? This ext number should be after the line is transfered, not the ext line (dont want the est line to ring on a group) just an ext after 198 picks up and the called transfered to the ext. How is this done?

Like having a group line dial multi lines at one time can you do the same thing with an extion number? someone calls in 198 picks uo the call they transfer to an extion number 105. can you make 105 ring a group of internal lines? i will also post this in IVM....

one item to clearify the call is coming in on an fxo line so it would not show on a second line as its the same call. i would have to check with the phone itself. one is a gx2000 grandstream.

when i am on a call i hear the call waiting tone for another call, does anyone know how you would pick up this call?

This worked i created a commond delimited file with the ext number assigned and it works great.

could i create a list under limit acceptable values and put the ext numbers in it?

two things where is this done in IVM? and what about if the ext are not concurent? ie... ext 100 to 115 then 200 and 300 and 400.

Is there a way to pull the call from voice mail as the customer is leaving a message?

when a client calls in and dont dial the correct ext# the call gets dropped. is there a way to have the system say things like that is not a current ext please retry. or is there a way to have it repeat the annoucment until that make the correct choice? yes there is a company directory but the bone heads that call in still cant seem to punch that right ext.

has there been any updates that have addressed this?