call transfer from a pstn to voip line using spa3102

[aguilar]

hello
i want to place a PSTN line just before a DR in order to keep the phone number the customer have already.
the call comes from a SPA3102 to a second DR which calls an extension (100) which makes an unconditional call to to my first voip line (5550252312) which in answred by the original DR and get this messages:

14:39:37.750 StratLink:nHangUp [CM104001] Call(83): Ln:10007@89959117 hung up call; cause: BYE; from IP:200.94.26.150
14:39:37.609 StratInOut:nCancel [CM104009] Call(84): Call from Ln:10007@89959117 to 800 has been terminated
14:39:35.187 MediaServerReporting:TMFhandler [MS211000] Call(84) Ln:10007@89959117: DTMF (RTP) from 200.94.26.152:31798 arrived. in-band DTMF tone detection is turned off.
14:39:35.156 MediaServerReporting:TMFhandler [MS211000] Call(83) Ln:10000@spa3102: DTMF (in-band) from 192.168.1.177:16452 detected.
14:39:26.015 CallLegImpl:nConnected [CM103001] Call(83): Created audio channel for Ln:10007@89959117 (200.94.26.152:31796) with Media Server (189.140.204.183:49929)
14:39:25.921 StratInOut:nConnected [CM104005] Call(83): Setup completed for call from Ln:10000@spa3102 to Ln:10007@89959117
14:39:25.875 CallLegImpl:nConnected [CM103001] Call(84): Created audio channel for Ln:10007@89959117 (200.94.26.152:31798) with Media Server (189.140.204.183:49931)
14:39:25.796 CallConf:nIncoming [CM103002] Call(84): Incoming call from 5589959117 (Ln:10007@89959117) to sip:[email protected]:5060
14:39:25.656 Endpoint::findSource [CM210001] Ambiguous configuration.2 source lines correspond to incoming call.INVITE Parameters: Contact IP: '200.94.26.150'; From: '5589959117'; To: '5550252312'; Rline: '5550252312'; Contact: '5589959117'. Line:CfgExtLine:10007 used as fallback!
14:39:25.546 Endpoint::findSource [CM210001] Ambiguous configuration.2 source lines correspond to incoming call.INVITE Parameters: Contact IP: '200.94.26.150'; From: '5589959117'; To: '5550252312'; Rline: '5550252312'; Contact: '5589959117'. Line:CfgExtLine:10007 used as fallback!
14:39:24.921 CallTarget::create [CM105002] Destination is overwritten by Forward all rule for Ext.100 to Number:050252312
14:39:10.750 CallLegImpl:nConnected [CM103001] Call(83): Created audio channel for Ln:10000@spa3102 (192.168.1.177:16452) with Media Server (192.168.1.84:7228)
14:39:10.718 CallConf:nIncoming [CM103002] Call(83): Incoming call from 10000 (Ln:10000@spa3102) to sip:[email protected]


your help will be greatly appreciated

 

[miraportuga]

No idea if this is possible, but, cant you just direct your SPA3102 to the original DR instead of using the second one? it would make things alot simpler...

Cheeers

 

[aguilar]

thanks for the reply
yes but i need to release release the pstn line for get more calls

regards

 

[miraportuga]

Dont know if this will do anything but have you tried already on your VSP Lines configuration(assuming that the provider is the same for all) under Other Options, to enable the option Use Ip in contacts and below input 200.94.26.150 .
Not having this option and the right address for my service gave me some headache before... try that... and tell me if it made any diference..

Cheers

 

[aguilar]

hi again
you mean in the PSTN line? i have just one all the others are voip lines from the same provider

regards

 

[miraportuga]

Voip Lines.... on the Voip Lines try what i told you to see if it works,

Cheers

 

[Mirzab]

I don't get that sequence. Even if you forward the PSTN call out on another line the original caller will still be on the PSTN line he called. The only way to "free up" this PSTN line would be for that inbound caller to call another line instead. Or am I missing something? Wouldn't be the first time :?

 

[miraportuga]

I was just thinking about that myself now too, all that call here call there, got me confused... lol :shock:

 

[aguilar]

yes is a little confuse but the reason for use the pstn number and release asap is for get another call, we are replacing and old conventional pbx and the users had a single number for 6 cascaded numbers
hope this explanation dont be more confused


regards

 

[miraportuga]

Thats not the issue here, the issue here, and you got me confused here, If the caller is calling to the PSTN line the line will never be released till the call ends, Even if you're calling to a Voip line right after, the PSTN line will always be busy becouse thats the number the Caller dialed. Only option is to add more PSTN lines and use a SPA400 for example or... get DID's and give ppl the new numbers. You can always set up a DR with a short message saying that people can also call to number x, y and someting else(this would be the Voip lines with Did number) and after that you direct them to the normal phone so that you can answer the call. this way the person that called will know she has more numbers avaible...

Cheers

 

[aguilar]

ok at this point, its clear for me that "If the caller is calling to the PSTN line the line will never be released till the call ends, " the point is just the oposite we want to give an UNIQUE number for call and the pbx redirect to the next available LINE
this number is already know by the customers and we dont want to give them a serie of voip numbers in our case are just 6 lines but what happen if this number goes to 50 or 100

once again thanks a lot for the advise

 

[Mirzab]

The solution used for many years has been multiple PSTN lines with a single primary number that "rolls over" to next available PSTN line. You can implement this solution with a multiple FXO gateway. If bandwidth is adequate there are business VoIP providers and DID providers that supply inbound numbers that can accommodate multiple streams. I use a toll free DID that will accommodate up to 10 simultaneous inbound calls on one number. This solution is much easier on the budget but will require a change of number. These are pretty much your options.

 

[aguilar]

thanks for the reply Mirzab,
actually i am answering a call with a SPA3102 using a DR for give the caller the new (voip) number and transfer the call to the second DR wich rolls over the next available VOIP line, but i was trying to optimise the precess

best regards

 

[sanyaolu]

Mirzab,

I belive what you are talking about is a PILOT number that clients will call and the PILOT is route the calls to the next free available inbound numbers that you have.

My sugestion will be to have your PSTN number as the PILOT # so people call that number. Then you set "Call forward on busy" to one of your DID Numbers which is attached and mapped to your SIP account available in your VoIP Lines. The call forward on busy will be done on your PSTN provider PBX hence, first call will be recieved on the PSTN and subsequent calls will be diverted to you DID numbers whenever your PSTN is busy

Then you can do whatever you want further with this solution.