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    Siproxd, setup and configuration for voip… works great!!!

    Scheduled Pinned Locked Moved pfSense Packages
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    • W
      wallabybob
      last edited by

      From the WEB GUI Services -> siproxd then click the Save button seems to restart siproxd.

      If you have ssh'd in or from the console (slight modification of a recipe from an earlier reply in this topic:

      pkill siproxd

      pgrep siproxd (should show up nothing here)

      siproxd -c /usr/local/etc/siproxd.conf

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      • J
        jigpe
        last edited by

        Awesome! Thanks :)

        But when i call soho router restarted..weird router…

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        • B
          blackb1rd
          last edited by

          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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          • D
            danswartz
            last edited by

            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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            • C
              cjkeeme
              last edited by

              To the original poster.  Thank you.  :)

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              • C
                cjkeeme
                last edited by

                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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                • B
                  bb-mitch
                  last edited by

                  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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                  • V
                    vronp
                    last edited by

                    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,

                    @bb-mitch:

                    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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                    • B
                      bb-mitch
                      last edited by

                      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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                      • V
                        vronp
                        last edited by

                        Well, this is interesting.  Both phones were at firmware level 5.2.  I upgraded both to 6.1.5 (latest) and now everything works !

                        @bb-mitch:

                        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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                        • B
                          bb-mitch
                          last edited by

                          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.

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                          • V
                            vronp
                            last edited by

                            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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                            • B
                              bb-mitch
                              last edited by

                              Doesn't look like you have flushed states to me.

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                              • V
                                vronp
                                last edited by

                                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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                                • B
                                  bb-mitch
                                  last edited by

                                  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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                                  • V
                                    vronp
                                    last edited by

                                    What version of pfsense are you using?

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                                    • V
                                      vronp
                                      last edited by

                                      Well, I turned on siproxyd and it all works now.

                                      BTW, this is 1.2.3.

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                                      • H
                                        hcoin
                                        last edited by

                                        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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                                        • T
                                          torontob
                                          last edited by

                                          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

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                                          • J
                                            jmwilson86
                                            last edited by

                                            I am having a similiar issue. We are Running ver 1.2.3. with 3 Fonality hosted phones, and a full T1. We are also running ntop, siproxd, and a few other packages. The issue is with call quality even at low bandwidth utlilization by other network devices.  We are using RTP 10000-20000 ports with firewall rule to allow traffic from IP address.(Fonality)  Fonality gave us a host range and we are still trying to figure out how to get a DNS name from them since we cant enter a range on PFsesne? Any ideas?

                                            We are getting the calls cut off for a few seconds every minute or two and not dropped.  We are loosing about 2-3 seconds of call quality ever 60-80 seconds.  Have ran traffic shapper till I am blue in the face and not sure what else to do!!  Any ideas.  See errors below from system log.

                                            Jan 24 16:58:37 siproxd[49015]: siproxd.c:287 INFO:siproxd-0.7.0-4577 i386-unknown-freebsd7.0 started
                                            Jan 24 16:58:37 siproxd[49015]: sock.c:65 INFO:bound to port 5060
                                            Jan 24 16:58:37 siproxd[49015]: siproxd.c:241 INFO:daemonized, pid=49015
                                            Jan 24 16:58:37 siproxd[49013]: siproxd.c:193 INFO:siproxd-0.7.0-4577 i386-unknown-freebsd7.0 starting up
                                            Jan 24 16:58:37 siproxd[49013]: readconf.c:309 ERROR:unknown keyword in config file, line:"load_plugin=plugin_logcall.la"
                                            Jan 24 16:58:37 siproxd[49013]: readconf.c:309 ERROR:unknown keyword in config file, line:"plugindir=/usr/local/lib/siproxd/"
                                            Jan 24 16:51:27 siproxd[20210]: dejitter.c:404 WARNING:stopping opposite stream
                                            Jan 24 16:51:27 siproxd[20210]: dejitter.c:397 ERROR:sendto() [74.115.98.40:13714 size=32] delayed call failed: Bad file descriptor
                                            Jan 24 16:32:41 check_reload_status: reloading filter

                                            also set my service curve to: 512=m1  5000=d  300=m2 for VOIP up and down.  Have made priority of 7 in parent q?  We have tried also just 300=m2 and no real difference for the voip ques.. I am beginning to think the issue is bc of my t1 but all seems ok there!

                                            Thanks for help and insight.  I am willing to look at any and all ways to fix.

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