Search results

It is not really helpful showing screenshots of incoming for one trunk and then posting incoming call for another trunk :) For your Siptalk inbound trunk use 1040xxx as the destination number. For the iiNet trunk, use 02xxxxxxxx format as the destination number. In any case just have one trunk...

I had this for a while, so I use IP address instead of host names for my gateway registrations. I suspect this condition is prevalent when using some routers.

That would be an issue with your set up then. I just tested my syntax on my Yealink T28 and it is working alert_info=http://example.com;info=alert-autoanswer;delay=0 May be post a screenshot of your dial plan block.

I am a bit confused....is your DID with iiNet or Siptalk? Your inbound route regex is not matching the correct incoming number. If your trunk is with Siptalk then your inbound should be matching the Siptalk extension number as the destination number.

You may need to adjust this a little bit depending on the phones in question:

I don't think calling Telstra will be of much help but see if you can escalate to level 2.

Ok....then Draytek is doing this by default. Try another router.

You have a few options, actually more than a few Change provider. Try aussiebroadband, Telecube (siptalk) etc Configure Draytek to remark all packets to DSCP 0 Get another router which is easier to configure Use g729 codec Ask Telstra to move you from DSCP mapped profile to Default profile

Did you configure Draytek firewall to remark all packets from Fusion to DSCP 0?

Correct; there can be two issues with Telstra NBN connections- the first one where Telstra gobbles up all non DSCP0 packets and the second where TC-1 bandwidth is limited to 150kbps and your connection profile has DSCP mapping. My script only checks the first one while the second is evident from...

Yes, indeed. It is not that simple to set DSCP. You need to remark every packet to DSCP 0 coming from Fusion IP using Draytek's firewall. You can run a test using my script: http://whrl.pl/RefWAK

Yes, I thought so :) Your connection through Telstra is on a DSCP profile which has only 150kbps for VoIP. Assuming one ALAW call uses around 85kbps, the second call would only work if you get at least 170kbps TC-1 traffic class. You have many options to solve the issue. Which router do you use?

I probably can help; but before getting into the details could you please state the country, ISP, type of internet and voice provider?

Try and see if this could work http://www.pbxforums.com/threads/phonebook-only-supports-yealink-currently.161/

Would mod_distributor work? https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor

There's an import button on the contacts page, but I am not sure if it is going to work properly. Probably test with a few contacts so undoing will be easier if things don't work out.

Check your mtu over 4G and packet size of the INVITE using siptrace.

I don't think you can sync from google contacts but it would be possible to export from google as csv and import into Fusion. I remember there was an issue with the import, not sure if it's been fixed recently.

If generous codec selection is available, the device originating the call gets to choose the codec based on the preference set on it. Now, if you set a single codec on the PBX and turn off transcoding, it should force the client to change the codec but you'll always run the risk of call setup...

Can I please request someone running MySQL/MariaDB to test this script and give me feedback? Thanks!