Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: Gautham on November 07, 2022, 02:44:11 PM
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Dear team,
I got a sip line which i like to configure directly to trunk. I register the details with xlite/Microsip it is working fine. But, not working with genesys trunk. I think I am missing something in trunk annex options.
Currently configured option:
[AuthClient]
password=xxxxxx
username=50806xxxxxxx
[TServer]
contact=sip:50806xxxxxx@ben.com:5060
enable-extension-headers=none
enforce-trusted=true
geo-location=
p-preferred-identity=sip:50806xxxxxxx@10.x.x.x:5060
prefix=0
record=true
refer-enabled=false
reuse-sdp-on-reinvite=true
ring-tone-on-make-call=true
sip-enable-100rel=false
sip-preserve-contact=true
subscription-id=Environment
username=50806xxxxxxx
xlite account details which is working:
Display name: 50806XXXXXXX
Username:50806XXXXXXX
password:XXXXXX
Authorization username:50806XXXXXXX
domain: ben.com
Checked the register with domain option
send outbound via proxy -->10.x.x.x
please help me to configure the trunk.
Thanks,
Goutham
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And what does logs say?
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Had a joined session with service provider and found that the header is not matching with the requirement. Need to modify the from and to address in the invite. How we can modify the invite "from" and "to" address in Genesys
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Again sir... Without logs there is no exact way to help you
You can search the forum as well
Enviado de meu SM-N9600 usando o Tapatalk
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[quote author=Gautham link=topic=12362.msg55096#msg55096 date=1668187329]
Had a joined session with service provider and found that the header is not matching with the requirement. Need to modify the from and to address in the invite. How we can modify the invite "from" and "to" address in Genesys
[/quote]
It means the trunk is registered successfully. You have problems directly with the call. From and To fields in SIP invite are who dialed and whom dialed. They shouldn't match some format, per my knowledge(maybe the fields length). You need to trace your call in SIP Server logs to understand what is going on there.