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1

Other Yate server issues / Routing some calls by javascript module

« on: December 28, 2017, 05:08:52 AM »

Hello Yate team

I want to route incomming calls by asking a separate web service for the destination.
I will write a script for instance named inboundjsrouting.js
I think the javascript module will be the best solution.
But I don't want route all of my calls using the javascript module
The most calls, I want to route by regexroute.conf (how it works currently)
Therefore I think I can do a pre-routing , for instance all calls which are comming in by the number +49123456987 in the section contexts in regexroute.conf to a separate context for instance jsrouting in which I can call the javascript module.
Is this the right way?
And how I have to call the javascript module and the script (in the example  inboundjsrouting.js) in the separate context (in the example jsrouting) in the regexroute.conf

Thanks for your help
Burkhard

2

Yate users hangout place / Fallback only if SIP response 408 TIMEOUT

« on: August 29, 2017, 06:17:51 AM »

Hello

in the doku of the regexroute.conf is described a round robin example with a fallback route.
Is it possible to call the fallback route only if one defined SIP response code came back from the regular route?
For instance:
I make a call through a regular route and I get back the SIP response code 408 TIMEOUT because the destination is down.
Then it should be used the fallback route.
If I make a call through the regular route and I get back the SIP response code 487 USER BUSY, then it shall not be used the fallback route.
Is there a variable where the SIP response code can used in the regexroute dialplan ?

Thank you for help
Kind regards
Burkhard

3

Other Yate server issues / Record a call in regexroute

« on: July 04, 2017, 04:50:49 AM »

Hello
is there a way to record a call in the regexroute.conf

.*49351898818077$=wave/record//var/records/inbound_${called}_${caller}$(timestamp).2slin
.*49351898818077$=sip/sip:+49302555880114778@213.XX.XX.XX:6045
This creates a valid record but it dials not the following destination

.*49351898818077$=sip/sip:+49302555880114778@213.XX.XX.XX:6045
.*49351898818077$=wave/record//var/records/inbound_${called}_${caller}$(timestamp).2slin
This dials the destination, but the record is empty

The call flow shoud be
Caller => yate where the call will recorded => PBX where the call is distributed to the callee

I will be happy  for any explanation, or better, examples
Thanks
Burkhard

4

Other Yate server issues / Routing calls beginning wiht +

« on: February 02, 2016, 07:51:31 AM »

Hi

I want to route some calls, beginning with +4935121324 to an other context than the other calls.
Therefore I wrote this entrys in regexroute.conf

[contexts]

 ^+4935121324\(.*\)=megacalltest
.*=inbound

But every call, which comes in, uses the default route to context inbound.
I don't understand why.

I have tryed

this
 ^\+\4935121324\(.*\)=megacalltest
.*=inbound

and

^\(+\)4935121324\(.*\)=megacalltest
.*=inbound

and

^\(+\)4935121324$.*=megacalltest
.*=inbound

The contexts inbound and megacalltest are existing.

Thanks for any advice

Kind Regards
Burkhard

5

Other Yate server issues / Round Robin in regexroute.conf

« on: January 07, 2016, 05:13:12 AM »

Hello

I dont understand why this
^.*$=fork $(index,$idx00,|sip/sip:\0@62.XXX.XX.XX0:5065,|sip/sip:\0@213.XX.XX.XX0:5065)
not works

It should distribute all incomming calls to more than one telephonie systems

I followed your example in the documentation
http://docs.yate.ro/wiki/Round_Robin_Routing

If I dont use round robin, like this
.*$=sip/sip:\0@62.XXX.XX.XX0:5065
it works fine

Here the Log
20160107114728.768263 <sip/2:ALL> YateSIPConnection::YateSIPConnection(0x6e69800658f0,0x6e6980065090) [0x6e6980065d40]
20160107114728.768731 <sip/2:ALL> NAT address is '(null)' [0x6e6980065d40]
20160107114728.769961 <sip/2:ALL> Set media: audio=alaw,mulaw [0x6e6980065d40]
20160107114728.771294 <INFO> Classifying caller '+493512XXXXX' in context 'inbound' in 273 usec
20160107114728.771470 <cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/2'
20160107114728.772071 <INFO> Routing call to '+493518XXXXXX' in context 'inbound' via 'fork |sip/sip:+493518XXXXXX@213.XX.XX.XX0:5065' in 355 usec
20160107114728.772146 <sip/2:ALL> NAT address is '(null)' [0x6e6980065d40]
20160107114728.772928 <callfork:MILD> Call 'sip/2' ignoring modifier 'sip/sip:+493518XXXXXX@213.XX.XX.XX0:5065'
20160107114728.772989 <sip/2:ALL> YateSIPConnection::disconnected() '(null)' [0x6e6980065d40]
20160107114728.773215 <sip/2:MILD> Call rejected error='(null)' reason='(null)' [0x6e6980065d40]
20160107114728.775265 <sip/2:ALL> YateSIPConnection::hangup() state=0 trans=0x6e6980065090 error='failure' code=500 reason='(null)' [0x6e6980065d40]
20160107114728.775379 <sip/2:ALL> YateSIPConnection::~YateSIPConnection() [0x6e6980065d40]
20160107114728.775687 <sip:INFO> 'udp:0.0.0.0:5060' sending code 500 0x3927af0 to 85.XXX.XXX.206:5060 [0x390ddb0]

or

20160107111057.003374 <sip/1:ALL> YateSIPConnection::YateSIPConnection(0x6e6980058090,0x6e6980059f90) [0x6e698005a480]
20160107111057.003981 <sip/1:ALL> NAT address is '(null)' [0x6e698005a480]
20160107111057.005136 <sip/1:ALL> Set media: audio=alaw,mulaw [0x6e698005a480]
20160107111057.006194 <INFO> Classifying caller '+493512XXXXX' in context 'inbound' in 236 usec
20160107111057.006602 <cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/1'
20160107111057.007068 <INFO> Routing call to '+493518XXXXXX' in context 'inbound' via 'fork |sip/sip:+493518XXXXXX@62.XXX.XX.XX0:5065' in 340 usec
20160107111057.007241 <sip/1:ALL> NAT address is '(null)' [0x6e698005a480]
20160107111057.008891 <callfork:MILD> Call 'sip/1' ignoring modifier 'sip/sip:+493518XXXXXX@62.XXX.XX.XX0:5065'
20160107111057.008921 <sip/1:ALL> YateSIPConnection::disconnected() '(null)' [0x6e698005a480]
20160107111057.009072 <sip/1:MILD> Call rejected error='(null)' reason='(null)' [0x6e698005a480]
20160107111057.009426 <sip/1:ALL> YateSIPConnection::hangup() state=0 trans=0x6e6980059f90 error='failure' code=500 reason='(null)' [0x6e698005a480]
20160107111057.009464 <sip/1:ALL> YateSIPConnection::~YateSIPConnection() [0x6e698005a480]
20160107111057.010929 <sip:INFO> 'udp:0.0.0.0:5060' sending code 500 0x3925ba0 to 85.131.161.206:5060 [0x390ddb0]

Thank you for your advice
Regards
Burkhard