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TekSIP - General issues
24.07.2009 03:54:44
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We have IPBX that supports SIP Trunks and SIP Extensions. Up to now we have to use a SIPerator to make these connections. The IPBX does not handle the NAT or PAT properly. The IPBX uses the REFER method for call setup and will not work with REINVITE. This means call transfers and conferencing do not work without the SIPerator to translate between the private IPBX and the public SIP provider. We have a test SIP trunk from a provider available to us. Our IPBX is at 172.16.1.22 and uses port 5060. The TekSIP is installed at 172.16.1.88 and our outside IP is 70.79.13.12. The SIP provider is at 24.24.24.10 and uses port 15621. We need to authenticate to the SIP Provider.
Will the TekSIP work instead of the SIParator if the SIP Provider uses the same SIP protocols as our IPBX?
We do not know how to configure the TekSIP. Any examples would give us a good start. We have no problems setting up trunks and extensions on the same network, but do not understand the configuration definitions used in the TekSIP application. What should be defined as an endpoint, routings, and settings.
Any help is greatly appreciated. GFM
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24.07.2009 11:14:54
 Admin Administrator Posts: 254
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Hi,
You need to define an endpoint for the IPBX at Endpoint tab and a routing (A default route for instance) entry at routing tab:

TekSIP supports NAT traversal. You can use UPnP or static external IP address.
Best regards,
Yasin KAPLAN edited by admin on 24.07.2009
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28.07.2009 19:58:10
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Thanks. We will let you know how it works.
GFM
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