Yate server > Yate bugs

[fix] Yate dialing problems...

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noci:
(either a bug in software, or in documentation)
I have: a few Voip Phones, Gigaset DX800 & S450-IP... the connect to yate... (no issue there).
(the connect to accounts like ext1@mydomain ..ext8@mydomain )
I have: 3 voip accounts with ISP1 and 2 with ISP2.    All accounts are basically phone accounts not gateways.
(LINE_11=phn1@isp1.domain, ... LINE_13=phn3@isp1.domains) & (LINE_21=phn1@isp2.domain & LINE_22=phn2@isp2.domain)

To dial out i need the SIP packet to have (From: phn1@isp1.domain; to: dialednumber@isp1.domain)

but the Calls generated by regex:
^012345678$=line/\1;line=LINE_11
does Become:     
SIP( From: ext1@mylocalhostname; To:dialednumber@isp1.domain)   which is halfway good....

^012345679$=line/\1;line=LINE_12;caller=phn2
does Become:
SIP( From: phn1@mylocalhostname; To:dialednumber@isp1.domain)   which is 3/4 good...
But is still rejected by the isp1..., due to the from.

dialing using the Sip extention  does n't work out as there is no way to select the right line:
^34567$=sip/sip:\1@isp1.domain    # is still missing a way to select a specific line...)
when using:
^34567$=sip/sip:\1@isp1.domain;caller=phn1    # is still missing a way to select a specific line...)
and delivers:
SIP( From: phn1@mylocalhostname; To:dialednumber@isp1.domain)   which is also 3/4 good...

If there is a right way it surely is hidden good in docs, site, ...
So i hope someone here can help out.

marian:
You may always specify the line to use when routing to a sip target:
^34567$=sip/sip:\1@isp1.domain;line=MY_LINE

You may set the from and to headers as you wish to (if needed):
^34567$=sip/sip:\1@isp1.domain;line=MY_LINE;osip_From=MY_FROM_URI;osip_To=MY_TO_URI

noci:
The first form was documented... but rejected by the ISP as the From url was the URL used for connecting the Phone -> Yate.

Thanks for the 2nd form, using the osip_From worked..., but i haven't found that in the documentation... It might need a bit better examples/ being referenced.
This means a lot of dial entries are needed depending on line chosen. + info that is IN the accfile.conf needs to be dupplicated to regexfile.conf.

Why not have an option to use the info from accfile.conf (or database profile) or registration info of the phone .. or maybe even the incoming dial info on connect through (if the ISP supports that, it wont work for me though).

noci:
The osip_From:sip:mynumber@isp;   entry worked.

marian:
You may set parameters for outbound calls in accfile for sip accounts by using the prefix 'out':
out:osip_From=
out:osip_To=
Inbound call parameters may be set using the prefix 'inb:'

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