d1rage5

August 20, 2008

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....

August 18, 2008

Dear community,

I've read this forum up and down and tried everything, but I'm not able to setup skype with uplink. The following things are working:

- SIP account is correct configurated in Uplink

- Uplink does connect to skype

When I receive a call uplink says "incoming call" but nothing happens in skype.

When I want to make a call skype warns me that I haven't got enough cash(Well this is probably because I haven't got a SkypeOut account).

I tried almost everything, installed axon, put the right prefixes before. Started Sykpe first, started uplink first, started skype detached uplink, runned uplink, and much more. I'm gettin totally nuts. Any idea would be very appreciated.

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.

August 18, 2008

hey;
did anyone ever respond to you on this or did u find the answer?

thanks

Michael

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.

August 18, 2008

who is your voip? I am looking for something like this and am not sure of different people for sip outgoing?
thanks

Michael

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

August 18, 2008

hi;
so how did u get it to work?

thanks

Michael

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...

August 7, 2008

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.

July 29, 2008

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?

July 19, 2008

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?

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)

13:51:53 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)

13:51:57 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

July 19, 2008

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?

July 8, 2008

I think the best way is to use a plug-in with IVM. The plug-in would reference a database to see which ring group is associated with that extension, and then it would return this to IVM as a variable which IVM could use in the next OGM to transfer to.

E.g.

Internal employee dials 105. Extension 105 rings.

Outside caller calls in and enters extension 105 (or some kind of reference number like employee id). IVM references the database and determines that 702 is the correct ring group for that number.

IVM transfers to 702, and extension 105 along with a couple of other extensions ring.

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

July 8, 2008

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.

July 7, 2008

Sorry for the misunderstanding, I'm having trouble figuring out what you're trying to accomplish.

In axon you can select the group or extension you want each external line to ring for an incoming call. I assume by 'I do not want the main call to RING on a group of numbers' that you will set it to ring a particular extension.

This is the part that confuses me, 'Can that extension dial a group after being routed to the extension?' and 'just the extension ring a group after being routed to that ext.'

Isn't this the same as simply routing to a group in the first place?

If I can better understand what you are trying to do, I'm sure I can come up with a solution for you, but at the moment I am at a loss to the purpose of all this.

Is it that you want to be able to call in an extension directly, and then let callers dial an extension which actually dials a group of extensions? Are you trying to differentiate between calls made directly to an extension vs calls sent there (e.g. by an IVM transfer)??

Please provide further details.

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)?

July 7, 2008

Well for one thing, the group number needs to be different to the extension number (how else would Axon be able to tell whether you wanted to call just 1 extension or the whole lot). Typically group numbers start with 7, and extensions with 1. However if most or all of the time you want to ring more than 1 extension when calling in, you could choose to switch this so that by default when you call 105, it calls the group of extensions related to 105 including 105, and when you call 705 it rings the sole extension directly.

I'm not exactly sure what determines which voice mail is used by default when there are multiple extensions in a group.

I would think (from a programmer's perspective) that it's either:

1) The mailbox of the first extension in the group list

2) The extension which times out first (has the shortest cut-off before voice mail kicks in)

If you can't get it to go to the right mailbox, you can use IVM's confirmed transfer feature. If the call transfer is not confirmed (i.e. not picked up in time), you could set it to go to a specific extension's mailbox.

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?

July 6, 2008

I answered your post in the IVM forum. Next time, please try to stick to one forum... it keeps things organized and helps keep related discussion in the same topic rather than across different forums.

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.

July 6, 2008

Yes, when using IVM in conjunction with Axon, a ring group/queue can be created and used to ring multiple internal extensions at a time.

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.

July 6, 2008

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?

July 6, 2008

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....

July 1, 2008

It depends on the phone you're using. Unlike analogue telephony, VoIP phones don't use tones or flash hooks to signal switch lines, instead the phone manufacturer will develop their own method to switch between phone lines and then use the standard SIP protocol to relay this information to Axon / the PBX.

For some IP phones and soft phones there are 'line appearance' buttons which allow you to switch between virtual 'lines' as needed (e.g. when a call waiting call comes in, the 2nd line appearance button flashes, and user can switch to the call waiting call by pressing the flashing button). On other devices, the old 'flash hook' (hanging-up and quickly picking up again) can trigger the device to relay a line switch to Axon, or sometimes there will be a clearly labelled 'Flash' or 'Link' button on the phone itself you can use to switch calls.

Note 1: Not all IP phones or devices support multiple lines/call waiting (in VoIP telephony, Call Waiting is the same as having a 2nd line appearance for incoming calls)

Note 2: Some IP devices need to have their Call Waiting feature enabled before it can be used

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.

July 1, 2008

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?

July 1, 2008

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

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

July 1, 2008

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.

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

July 1, 2008

If you're using IVM to transfer calls to the extension number they put in, you can limit the accepted data values (e.g. from 100 to 110) and any numbers outside of that range will automatically trigger IVM to say the number was invalid and to try again (repeating the OGM)

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.

July 1, 2008

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

July 1, 2008

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.

June 29, 2008

Unfortunately this is one set-back of the Uplink software. Currently there is no way to receive the digit inputs through Uplink from Skype. The best you can do is to have Uplink calls go to another IVM extension/ogm that answers without asking for digit input and just transfers directly to an extension.. not really a solution though.

has there been any updates that have addressed this?

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d1rage5

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Posts posted by d1rage5

(Definity + Vega + Axon + SIP2Skype) problem

!!!! Uplink says Incoming Call, Skype does nothing !!!!

Step by step configuration of AXON and Uplink for SIP phone which is running remotely to call Skype

Outbound VoIP provider multi-line

extions that dont exist

IVM Fax Transfer

bandwidth throtting

incoming call from voip phone to same system

incoming call from voip phone to same system

can an ext number dial multi lines at one time

password on phone when you connect to voice mail

can an ext number dial multi lines at one time

can an ext number dial multi lines at one time

can an ext number dial multi lines at one time

can an ext number dial multi lines at one time

can an ext number dial multi lines at one time

can an ext number dial multi lines at one time

call waiting

call waiting

extions that dont exist

extions that dont exist

extions that dont exist

pick up call from voice mail

extions that dont exist

skype call to SIP (AXON)

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