FreePBX inbound call issue
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There are only two occasions when you want to port forward the traffic for your voice over IP.
Condition one if you have external phones.
Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.
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@jaredbusch said in FreePBX inbound call issue:
There are only two occasions when you want to port forward the traffic for your voice over IP.
Condition one if you have external phones.
Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.
My SIP provider does actually use IP validation instead of registration.
I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.
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@samsmart84 said in FreePBX inbound call issue:
@jaredbusch said in FreePBX inbound call issue:
There are only two occasions when you want to port forward the traffic for your voice over IP.
Condition one if you have external phones.
Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.
My SIP provider does actually use IP validation instead of registration.
I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.
Actually no that’s not what was happening before. Before when the inbound calls were not working nothing hit the PBX
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@jaredbusch said in FreePBX inbound call issue:
@samsmart84 said in FreePBX inbound call issue:
@jaredbusch said in FreePBX inbound call issue:
There are only two occasions when you want to port forward the traffic for your voice over IP.
Condition one if you have external phones.
Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.
My SIP provider does actually use IP validation instead of registration.
I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.
Actually no that’s not what was happening before. Before when the inbound calls were not working nothing hit the PBX
And it still fails to show anything on the PBX when it comes up blank.. but Sophos shows it as being pushed to the PBX with no drops! How the heck am I supposed to troubleshoot that?
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@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:
There are only two occasions when you want to port forward the traffic for your voice over IP.
Condition one if you have external phones.
Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.
My SIP provider does actually use IP validation instead of registration.
I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.
Actually no that’s not what was happening before. Before when the inbound calls were not working nothing hit the PBX
And it still fails to show anything on the PBX when it comes up blank.. but Sophos shows it as being pushed to the PBX with no drops! How the heck am I supposed to troubleshoot that?
With the packet capture on up near port of the port going to the PBX
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So I messed with my SIP trunk settings and inbound calling changed from dead silence to a busy signal so it's definitely getting through the firewall.
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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
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@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 -
@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=ulawYeah 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.
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@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=ulawYeah 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?
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@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=ulawYeah 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
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@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=ulawYeah 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.
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@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=ulawYeah 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
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@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=ulawYeah 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.
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@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=ulawYeah 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
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@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=ulawYeah 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.
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@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=ulawYeah 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
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Another vote for FreePBX 14.
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@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/
