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Messages - bnaetsch

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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

Yate users hangout place / Fallback only if SIP response 408 TIMEOUT
« on: August 29, 2017, 06:17:51 AM »

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

Other Yate server issues / Re: Record a call in regexroute
« on: July 05, 2017, 03:15:53 AM »
Thank you marian.

your example works fine, I get 2 voice files and the call is connected to the destination.

King regards

Other Yate server issues / Record a call in regexroute
« on: July 04, 2017, 04:50:49 AM »
is there a way to record a call in the regexroute.conf

This creates a valid record but it dials not the following destination

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

Other Yate server issues / Re: Routing calls beginning wiht +
« on: February 09, 2016, 09:18:23 AM »
Thank you
It works fine

Other Yate server issues / Re: Routing calls beginning wiht +
« on: February 05, 2016, 04:05:32 AM »
Thank you for your advice, the change of the context works now.

I have a general question

The yate server must work as a proxy, like this

Yate get calls from different servers without registration but with authentication.
Yate has to register on another PBX

The call from the different servers comes in and will routed to the PBX where yate has registered itself.

So I made 2 accounts in accfile.conf
One account for the incomming calls without registration

description=Testaccount for Incomming

One account for the registering to the  PBX

description=Testaccount fuer Outgoing

For tests, I use a defined phone number range like this +4935121324241XX
In the past I worked with asterisk, but I think yate is smaller and also more performant
I configured in the file regexroute.conf the following context.

${called}^+4935121324241.*S=echo called is ${called}

Is this the right way ?

Many Thanks

Other Yate server issues / Routing calls beginning wiht +
« on: February 02, 2016, 07:51:31 AM »

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



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

I have tryed






The contexts inbound and megacalltest are existing.

Thanks for any advice

Kind Regards

Other Yate server issues / Re: Round Robin in regexroute.conf
« on: January 08, 2016, 02:22:45 AM »
Thanks for your advice

it works fine


Other Yate server issues / Round Robin in regexroute.conf
« on: January 07, 2016, 05:13:12 AM »

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

If I dont use round robin, like this
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:' sending code 500 0x3927af0 to 85.XXX.XXX.206:5060 [0x390ddb0]


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:' sending code 500 0x3925ba0 to [0x390ddb0]

Thank you for your advice

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