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    FreePBX inbound call issue

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    • JaredBuschJ
      JaredBusch @SamSmart84
      last edited by JaredBusch

      @samsmart84 said in FreePBX inbound call issue:

      @jaredbusch said in FreePBX inbound call issue:

      @samsmart84 said in FreePBX inbound call issue:

      Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

      https://www.voip-info.org/asterisk-sip-qualify/

      Interesting that this was working before without requiring this

      Wow, that trunk is fucked up if you did not have those set...
      I am surprised shit ever worked.

      This is a typical SIP trunk setup.

      username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

      Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

      I am sure you have mentioned it in one post or another, but what version of what are you on?

      S 1 Reply Last reply Reply Quote 1
      • S
        SamSmart84 @JaredBusch
        last edited by SamSmart84

        @jaredbusch said in FreePBX inbound call issue:

        @samsmart84 said in FreePBX inbound call issue:

        @jaredbusch said in FreePBX inbound call issue:

        @samsmart84 said in FreePBX inbound call issue:

        Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

        https://www.voip-info.org/asterisk-sip-qualify/

        Interesting that this was working before without requiring this

        Wow, that trunk is fucked up if you did not have those set...
        I am surprised shit ever worked.

        This is a typical SIP trunk setup.

        username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

        Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

        I am sure you have mentioned it in one post or another, but what version of what are you on?

        It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

        JaredBuschJ 1 Reply Last reply Reply Quote 0
        • JaredBuschJ
          JaredBusch @SamSmart84
          last edited by

          @samsmart84 said in FreePBX inbound call issue:

          @jaredbusch said in FreePBX inbound call issue:

          @samsmart84 said in FreePBX inbound call issue:

          @jaredbusch said in FreePBX inbound call issue:

          @samsmart84 said in FreePBX inbound call issue:

          Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

          https://www.voip-info.org/asterisk-sip-qualify/

          Interesting that this was working before without requiring this

          Wow, that trunk is fucked up if you did not have those set...
          I am surprised shit ever worked.

          This is a typical SIP trunk setup.

          username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

          Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

          I am sure you have mentioned it in one post or another, but what version of what are you on?

          It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

          Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

          S 1 Reply Last reply Reply Quote 1
          • S
            SamSmart84 @JaredBusch
            last edited by

            @jaredbusch said in FreePBX inbound call issue:

            @samsmart84 said in FreePBX inbound call issue:

            @jaredbusch said in FreePBX inbound call issue:

            @samsmart84 said in FreePBX inbound call issue:

            @jaredbusch said in FreePBX inbound call issue:

            @samsmart84 said in FreePBX inbound call issue:

            Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

            https://www.voip-info.org/asterisk-sip-qualify/

            Interesting that this was working before without requiring this

            Wow, that trunk is fucked up if you did not have those set...
            I am surprised shit ever worked.

            This is a typical SIP trunk setup.

            username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

            Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

            I am sure you have mentioned it in one post or another, but what version of what are you on?

            It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

            Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

            Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

            JaredBuschJ 1 Reply Last reply Reply Quote 0
            • JaredBuschJ
              JaredBusch @SamSmart84
              last edited by

              @samsmart84 said in FreePBX inbound call issue:

              @jaredbusch said in FreePBX inbound call issue:

              @samsmart84 said in FreePBX inbound call issue:

              @jaredbusch said in FreePBX inbound call issue:

              @samsmart84 said in FreePBX inbound call issue:

              @jaredbusch said in FreePBX inbound call issue:

              @samsmart84 said in FreePBX inbound call issue:

              Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

              https://www.voip-info.org/asterisk-sip-qualify/

              Interesting that this was working before without requiring this

              Wow, that trunk is fucked up if you did not have those set...
              I am surprised shit ever worked.

              This is a typical SIP trunk setup.

              username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

              Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

              I am sure you have mentioned it in one post or another, but what version of what are you on?

              It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

              Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

              Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

              You are on Asterisk now, so stay on it.

              Move to FreePBX 14.

              S JaredBuschJ 2 Replies Last reply Reply Quote 1
              • S
                SamSmart84 @JaredBusch
                last edited by

                @jaredbusch said in FreePBX inbound call issue:

                @samsmart84 said in FreePBX inbound call issue:

                @jaredbusch said in FreePBX inbound call issue:

                @samsmart84 said in FreePBX inbound call issue:

                @jaredbusch said in FreePBX inbound call issue:

                @samsmart84 said in FreePBX inbound call issue:

                @jaredbusch said in FreePBX inbound call issue:

                @samsmart84 said in FreePBX inbound call issue:

                Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

                https://www.voip-info.org/asterisk-sip-qualify/

                Interesting that this was working before without requiring this

                Wow, that trunk is fucked up if you did not have those set...
                I am surprised shit ever worked.

                This is a typical SIP trunk setup.

                username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

                Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

                I am sure you have mentioned it in one post or another, but what version of what are you on?

                It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

                Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

                Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

                You are on Asterisk now, so stay on it.

                Move to FreePBX 14.

                Sounds like a plan! Thanks

                1 Reply Last reply Reply Quote 0
                • JaredBuschJ
                  JaredBusch @JaredBusch
                  last edited by

                  @jaredbusch said in FreePBX inbound call issue:

                  @samsmart84 said in FreePBX inbound call issue:

                  @jaredbusch said in FreePBX inbound call issue:

                  @samsmart84 said in FreePBX inbound call issue:

                  @jaredbusch said in FreePBX inbound call issue:

                  @samsmart84 said in FreePBX inbound call issue:

                  @jaredbusch said in FreePBX inbound call issue:

                  @samsmart84 said in FreePBX inbound call issue:

                  Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

                  https://www.voip-info.org/asterisk-sip-qualify/

                  Interesting that this was working before without requiring this

                  Wow, that trunk is fucked up if you did not have those set...
                  I am surprised shit ever worked.

                  This is a typical SIP trunk setup.

                  username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

                  Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

                  I am sure you have mentioned it in one post or another, but what version of what are you on?

                  It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

                  Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

                  Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

                  You are on Asterisk now, so stay on it.

                  Move to FreePBX 14.

                  I mean if you want to learn more, you could try Wazo or some other Asterisk distro.

                  But that is for people that want to be PBX people.

                  S 1 Reply Last reply Reply Quote 1
                  • S
                    SamSmart84 @JaredBusch
                    last edited by

                    @jaredbusch said in FreePBX inbound call issue:

                    @jaredbusch said in FreePBX inbound call issue:

                    @samsmart84 said in FreePBX inbound call issue:

                    @jaredbusch said in FreePBX inbound call issue:

                    @samsmart84 said in FreePBX inbound call issue:

                    @jaredbusch said in FreePBX inbound call issue:

                    @samsmart84 said in FreePBX inbound call issue:

                    @jaredbusch said in FreePBX inbound call issue:

                    @samsmart84 said in FreePBX inbound call issue:

                    Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

                    https://www.voip-info.org/asterisk-sip-qualify/

                    Interesting that this was working before without requiring this

                    Wow, that trunk is fucked up if you did not have those set...
                    I am surprised shit ever worked.

                    This is a typical SIP trunk setup.

                    username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

                    Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

                    I am sure you have mentioned it in one post or another, but what version of what are you on?

                    It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

                    Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

                    Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

                    You are on Asterisk now, so stay on it.

                    Move to FreePBX 14.

                    I mean if you want to learn more, you could try Wazo or some other Asterisk distro.

                    But that is for people that want to be PBX people.

                    Yeah I mainly just want something simple and stable at this point

                    1 1 Reply Last reply Reply Quote 0
                    • scottalanmillerS
                      scottalanmiller
                      last edited by

                      Another vote for FreePBX 14.

                      1 Reply Last reply Reply Quote 0
                      • 1
                        1337 @SamSmart84
                        last edited by 1337

                        @samsmart84 said in FreePBX inbound call issue:

                        But that is for people that want to be PBX people.

                        Yeah I mainly just want something simple and stable at this point

                        I'll put in a vote for 3CX then.

                        Easy to use, professional looking do-it-all web GUI. Very easy to install, good user forum. Free license for small installations.

                        We run it on a linux VM and it has been working great. 3CX also have good client software for Windows/Mac/Android/iOS that integrates well with the PBX. Otherwise we use Yealink phones.

                        Here is the page where you find the linux stuff:
                        https://www.3cx.com/phone-system/asterisk/

                        1 Reply Last reply Reply Quote 0
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