FreePBX inbound call issue
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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 I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.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/
