Siproxd, setup and configuration for voip… works great!!!
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Good afternoon :) Thanks for the tips! :) I have a question, how to restart the siproxd on ssh? Command not found "siprox -d restart"
jigp
1.2.2 -
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Awesome! Thanks :)
But when i call soho router restarted..weird router…
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siproxyd works great, but does someone know how to get it's packets through the Traffic Shaper? Or should I patiently wait for pfsense 2.0 :)
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You could try disabling the RTP proxy - dunno how much you like that idea (or whether it will work for you.) I ended up uninstalling siproxd for that (and other reasons), since I have only one client behind the pfsense - my asterisk server, so siproxd is not really needed.
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To the original poster. Thank you. :)
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siproxyd works great, but does someone know how to get it's packets through the Traffic Shaper? Or should I patiently wait for pfsense 2.0
I would also like to know if anyone knows the answer to this question. I have all my phones registered, but if someone is using too much bandwidth call quality goes down significantly.
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I recently added siproxd to a site that WAS working before adding it… I added it because of a single phone which is not "nat friendly" - special purpose phone (boardroom) - at any rate, on adding siproxd, I could get the Linksys phones to work (SPA942, SPA962) however the latest phone Cisco/Linksys SPA525G with 7.2.5 firmware SEEMS to work however the ringer doesn't ring and if you answer there is no audio (indicating lack of RTP stream).
I couldn't see a way around the problem, so I hacked the package to change the firewall rule so that only phones in a given "Alias" are added to the proxy. I'd like to share that change, but I found one thing a little frustrating - unless I create a dummy rule in the firewall, the Alias does not seem to be parsed into a table (which causes my change to fail).
I also noticed siproxd seems to be behind - and yet the packages list does NOT indicate it is lacking a maintainer...
Basically I've solved my own issue, but think it would benefit others...
Thoughts? Thanks all!
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bb-mitch,
I have two SPA962 that are configured identically. I recently moved from a Cisco 5505 firewall to pfsense.
With siproxd setup I have one phone working but the other refuses to.
Just wondering if you ran into this problem with your Linksys phones.
thanks,
I recently added siproxd to a site that WAS working before adding it… I added it because of a single phone which is not "nat friendly" - special purpose phone (boardroom) - at any rate, on adding siproxd, I could get the Linksys phones to work (SPA942, SPA962) however the latest phone Cisco/Linksys SPA525G with 7.2.5 firmware SEEMS to work however the ringer doesn't ring and if you answer there is no audio (indicating lack of RTP stream).
I couldn't see a way around the problem, so I hacked the package to change the firewall rule so that only phones in a given "Alias" are added to the proxy. I'd like to share that change, but I found one thing a little frustrating - unless I create a dummy rule in the firewall, the Alias does not seem to be parsed into a table (which causes my change to fail).
I also noticed siproxd seems to be behind - and yet the packages list does NOT indicate it is lacking a maintainer...
Basically I've solved my own issue, but think it would benefit others...
Thoughts? Thanks all!
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depending on what you are connecting to you could try turning off the nat options on the server and the phones and possibly the qualify options on teh asterisk server - one of my associates says he had to do that to support some newer cisco phones 79xx something I think?
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Well, this is interesting. Both phones were at firmware level 5.2. I upgraded both to 6.1.5 (latest) and now everything works !
depending on what you are connecting to you could try turning off the nat options on the server and the phones and possibly the qualify options on teh asterisk server - one of my associates says he had to do that to support some newer cisco phones 79xx something I think?
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ALWAYS try the various firmwares ;-)
They normally fix one thing and break something subtle, but 6.1.5(a) included a lot of fixes.
cheers. -
I spoke too soon. One of them works but the other still does not. This view of states seems to indicate why but I'm not sure what will fix this. The .47 phone works but .49 does not.
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Doesn't look like you have flushed states to me.
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Yes, correct. After flushing states, the "bad" phone is the only one that rings now and there is no audio.
There is another piece missing here.
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If you removed siproxd / disabled it, and the phone that wasn't working now rings, that means the SIP is working with NAT off.
siproxd also has the ability to proxy the RTP - this has to be enabled too if you need rtp. There needs to be some documentation for this package I think. I believe I understand a bunch of it - and don't mind contributing, but who is the package maintainer?
There are options / fields in the package gui that do not seem to be implemented or that I don't understand?SIP does things like connects the phone, and handles signalling (on hook, off hook, ring, call waiting, etc.).
RTP carries the audio or video streams AFTER SIP is used to set them up / define them.If a phone rings without siproxd but doesn't carry audio I would think you have a mismatch in your settigns somewhere. But if you don't control the server you should be seeking some help with the people that do - they can probably tell you exactly what you should set to work with their server.
m/
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What version of pfsense are you using?
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Well, I turned on siproxyd and it all works now.
BTW, this is 1.2.3.
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It would be really nice for semi-graceful failover if the pfSense GUI would allow siproxd to specify virtual ips in addition for the incoming and outgoing interfaces as well as offering the native interface addresses.
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Hi Guys,
I am using multiple Asterisk servers to connect to multiple providers on the internet. I also have enpoints from outside connecting to these Asterisk servers.
Endpoints connecting from outside to one of the Asterisk servers I have work just find as I have NAT forward port 5060 and RTP ports to one Asterisk server.
However, only one of my Asterisk servers can connect to the provider outside. If I try to connect more than one then the others stop working.
Should Siproxd be the answer for both inbound and outbound SIP?
Here is a diagram of what I have:
-Asterisk A -Asterisk B -Asterisk C -Asterisk D–>pfsense1.2.3INTERNET<--Provider(s) AND <--Endpoints
Thanks