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    Yealink Call Quality

    where would i change to udp m8 ? not done that b4
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    Yealink Call Quality

    Thanks for your reply m8, rport is on and the phones are showing tcp in the registrations page ... What did your issue turn out to be, its driving me nuts
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    Yealink Call Quality

    This is definitely something to do with the new freeswitch version.. is everybody now using 1.10 or are you guys still using 1.8.7 ? i didnt have any of these issues on 1.8.7
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    Max Calls Spec

    No-one have any ideas ? a little guidance would be appreciated :-)
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    Max Calls Spec

    Hi, so we are moving away from our vps mainly due to costs of vm's with them and setting up our own data centre. The specs are we will have a 100m over 1Gb bearer to start with and a 3 node proxmox cluster and an external shared iscsi storage. Two if the 3 cluster nodes will have 1 VM with 60Gb...
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    Yealink Call Quality

    We are using PCMA and we was using g722 but it was worse with that active. Prompts seem to sound better using opus however the transposing when dialing out is alot especially as these are vm
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    Yealink Call Quality

    Has anyone found that using the latest freeswitch 1.10.3 and up with yealink phones that they are getting jitter speech and even simple things like calling the voicemail the prompts are sounding a bit jittery too ? we have 4 different installs where this seems to be happening and i cant seem to...
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    Jitter Buffer

    Hi Guys i have an install and i am getting jitter and break up on the outbound leg form the users site, inbound voice is ok. I suspect there is a network or broadband issue on the users site however to try and help with the issue how would i enable the jitter buffer on the dialplan and where...
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    Context Not Found on Valid Domain

    are the new inbound number presenting the same way ? ie national number or sip user etc, you need to include a sip trace and let us know what the actual number is so you can compare it to what's being received by the system.
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    Context Not Found on Valid Domain

    Check your access controls , the fusionpbx team changed how they work a few updates ago i had the same issue, ensure you put the full cidr in there from your isp
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    Multi routes

    Hi guys, i have a client with 2 companies on the 1 server using fusionpbx (latest) , they both have their own sip trunks on the system, whats the easiest way to ensure each comapny uses its own trunks without doing multi tenant ? Thanks Kurt
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    Yealink Contacts

    Hi, i have been banging my head against the wall now for days, is there any way at all to get the yealink phones to pull the contacts from fusionpbx ? i have added entries, provisioned a yealink phone, i see remote phonebook but its empty, can someone pleeaseeeee tell me howto set this up ...
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    Yealink Provision

    Hi, quick question, we use yealink rps to provision our phones, when setting up the extension and device in fusionpbx i only have a profile for t58v however the phone is a t58a , can i use the t58v provision file for a t58a ? Thanks
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    Call Center Distinctive Ring

    Tried that, makes no change at all ...
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    Phone Registers rejected by ACL domains

    Also ensure you add your trunk provider ip address to fail2ban jail.conf under ignore ip , i had awful issues before i did that, then all was good after that.
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    Call Center Distinctive Ring

    Hi Guys, i know i can change ring tone on a group, but is there a way to change the ring tone on a call centre agent ? it just does the standard internal ring tone and i would like it to do a different one, i use Yealink T54w or 53w Thanks
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    Help with dialling international

    Awesome that's worked fine, thanks for the help :-)
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    Help with dialling international

    Hi, i have a user that uses mainly softphone, his pc also has an address book that the numbers are in this format +44thenthenumber dropping the 0 , this is normal and works on his mobile but no matter what i se as the dialplan it fails to dial in the cli i get the following error [ERR]...
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    Site Groups

    So i have an installation to do that has multi sites all on one server, what i need is for each site to be limited to how many calls that site can make and recieve, say 2 in 2 out at once, however there will be one large sip pipe into the system with 60 channels ? is this possible ?
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    Show orginal CallerID when forwarding to Mobile Phone

    PS it has nothing to do with the caller id on your gateway ... as that should follow whats set on your extension...