› VitalPBX Community Support › General Discussion › [VOIPTALK] Does VitalPBX support outbound proxies?
- This topic has 9 replies, 2 voices, and was last updated 2 years, 2 months ago by
davej.
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- October 29, 2018 at 5:49 pm
I have setup my voip providers details in the peer parameters and outgoing calls work ok.
I am behind a NAT router and the connection by proxy is meant to stop the requirement of opening up ports on my router.
My connection below works for outgoing calls but when I look in the logs it seems that there is no reference to the outboundproxy. Is this supported by VitalPBX?
type=friend
username=XXXXXXXX
secret=123456
dtmfmode=rfc2833
host=voiptalk.org
outboundproxy=nat.voiptalk.org
port=5065
fromuser=XXXXXXX
fromdomain=voiptalk.org
usereqphone=yes
canreinvite=noThanks
0
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- October 29, 2018 at 7:37 pm
- November 2, 2018 at 9:00 am
Thanks for your reply.
I have looked everywhere for the solution to the outgoing proxy. I have found instructions for connecting to an Asterisk server as follows but with no luck
[general] register => 844XXXX:xxxxx:844XXXX@voiptalk/844XXXX [voiptalk] type=friend username=844XXXX secret= xxxxx dtmfmode=rfc2833 host=voiptalk.org ;Below is will be the context you will use to receive incoming calls in extension.conf context=voiptalk_incoming outboundproxy=nat.voiptalk.org port=5065 fromuser=844XXXX fromdomain=voiptalk.org usereqphone=yes canreinvite=no
The proxy never shows as connecting in the logs the only error I get is the following;
<————->
[2018-11-02 08:52:56] VERBOSE[1790] chan_sip.c: — (8 headers 0 lines) —
[2018-11-02 08:52:56] VERBOSE[1790] chan_sip.c:
<— SIP read from UDP:77.240.48.94:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK31eaf7c6;rport=5060;received=*.*.*.*
From: <sip:844******@voiptalk.org>;tag=as42496865
To: <sip:844******@voiptalk.org>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.d515
Call-ID: 7fe2e8a67c3b8d932acf5b0f13acae90@[fe80::20c:29ff:fe7d:a5d3]
CSeq: 124 REGISTER
WWW-Authenticate: Digest realm=”voiptalk.org”, nonce=”5bdc1086000125754b69626c521a0151d6338ee4d15b71ec”, stale=true
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0<————->
I have connected with a 3CX system no problem but prefer VitalPBX as its cheaper ?
0- November 2, 2018 at 6:07 pm
The outbound proxy parameter is for send outbound signaling through the Proxy, that means that only works for outgoing calls and does nothing to do with incoming calls.
To configure NAT settings, you can go to the SIP Settings form under Network Tab and configure your NAT settings with your External and local address.
I am attaching you an example of a trunk with voiptalk. Outgoing and Incoming calls working as expected
0- November 2, 2018 at 6:28 pm
Thanks for your indepth reply. It seems that you are avoiding the use of a proxy and therfore reliant on the port forwarding from the router.
By using an outbound proxy there is no need to use port forwarding and the possible security implications that in entails.
If this is the only way of making it work I may look into what is needed but it does mean a lot of forwarded ports.
Thanks
0- November 3, 2018 at 5:28 pm
Due I am using the visual mode, I set the outbound proxy under advanced Tab, however, the trunk is still working as expected. What is your issue with the Incoming cal? There is not sound in the calls?
0- November 5, 2018 at 8:29 am
I have recreated the Trunk with exactly the same settings as you have shown using my account/password and the phone as usual dials out with no problem and the sound can be heard ok from both directions.
The issue seems to be that I dont get any inbound calls. Do you have port forwarding set on your router or does your pbx sit on an external IP address?
Thanks
0- November 5, 2018 at 10:49 am
I think I have now found the problem as VitalPBX is now showing the outbound proxy in the logs but rather than connecting the remote server is offering options and therfore disconnects.
———- SIP HISTORY for ’49d2eaa1082c86065e49ab7117afe932@192.168.1.6:5060′
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c: * SIP Call
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c: 001. OBproxy Using peer obproxy nat.voiptalk.org
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c: 002. TxReqRel OPTIONS / 102 OPTIONS – OPTIONS
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c: 003. Rx SIP/2.0 / 102 OPTIONS / 403 Forbidden – no OBP
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c: 004. NeedDestroy Setting needdestroy because got OPTIONS response
[2018-11-05 10:37:08] DEBUG[1792] chan_sip.c:
———- END SIP HISTORY for ’49d2eaa1082c86065e49ab7117afe932@192.168.1.6:5060′Is there somewhere I can look to see in more depth what option I have not given the remote server?
Thanks
0- November 5, 2018 at 1:35 pm
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