![]() ![]() ![]() Repl-bytes=927 repl-fasttrack-packets=0 repl-fasttrack-bytes=0 ![]() Orig-fasttrack-packets=0 orig-fasttrack-bytes=0 repl-packets=3 Timeout=58m21s orig-packets=3 orig-bytes=1 347 Repl-fasttrack-bytes=0 orig-rate=0bps > /ip firewall connection print detail where connection-type=sipĠ SAC s protocol=udp src-address=192.168.0.100:5060 Repl-packets=3 repl-bytes=252 repl-fasttrack-packets=0 Orig-bytes=336 orig-fasttrack-packets=0 orig-fasttrack-bytes=0 Icmp-type=8 icmp-code=0 icmp-id=521 timeout=58m16s orig-packets=4 The implementation of the router node is:Ĭode: Select all > /ip firewall connection print detail where protocol=icmpįlags: E - expected, S - seen-reply, A - assured, C - confirmed, D - dying,Ġ S C s protocol=icmp src-address=192.168.0.100 dst-address=199.87.121.233 Our router ( router) is connected to both ISPs and also to the sip-client node (an Ubuntu 14.04 docker node that simulates a SIP client). They connect to the Internet via GNS3 NAT nodes (if you doesn't know how GNS3 works, just consider that the isp1 and isp2 nodes just behave as real ISPs routers). The isp1 and isp2 nodes simulate the two different ISPs. The screenshot of my GNS3 setup is above: So, I'm showing this simplified setup here. However, I run a much more simple setup using GNS3. I have a production setup somebit complicated. Network setup and detailed how to reproduce: Check the NAT table and run a sniffer in the router and you will see that the router is routing the package via the second ISP but it's still applying the old NAT rule (for the first ISP) instead of the correct NAT rule. Try to re-register the SIP client in the SIP server and you will see that no SIP message returns and the re-register fails Change the distance of the default routes so the second ISP will be the active route (smaller distance) Set up a SIP client (in the internal network) to register in an external SIP server and do the register Create proper default routes (static routes) for each ISP (the first ISP with the smaller distance) Plug a router to two different ISPs (each one giving you a different real IP) and to an internal network Note: We have tested some real Mikrotiks (mibspe) and run some simulations in GNS3 with routerosx86 Mikrotik virtual machines (chr). If we clean the NAT table or even reboot the router, everything is gonna be ok again.Ħ.38(mibspe),6.38.5(chr),6.39.3(mibspe),6.41(chr) Because of this, the SIP register messages cannot reach the SIP server and the SIP connection drops. When changing the default route from one ISP provider to the another one (manually, or because the ISP link goes down), the Mikrotik applies the wrong NAT rule. In our setup we have two ISP providers, a SIP client with a private IP, and we're using NATs (a different NAT for each ISP provider) with SIG ALG translation, aka SIP nat helper. The information here is from a customer using a Yealink T48G and does not get overwritten when phone is restarted.SIP client cannot re-register in the SIP server after switching ISP (different NAT). Replace: 1215$1 (replace 215 with whatever area code needed) You dial 7-digits and the phone will prepend a 1 plus area code in front for a total of 11 digits.įor example: For systems with 4 Digit Extensions, as long as the extensions start with a 2,3,4,5,6, or 7. Settings ->Dial Plan ->Replace Rule ->enter above into appropriate box ->Addįor example: For systems with 4 Digit Extensions, as long as the extensions start with a 2,3,4,5,6 or 7.Ĭonfiguring directly in the phone's web interface, to prepend an area code(s) to a dialed number, you would enter 7-x use the 1(area code)$1. You dial 10 digits and the phone will prepend a 1 in front for a total of 11 digits. Manual Configuration of a Yealink digit map/dial plan follow these directions: Prepending a "1" to callsĬonfiguring directly in the phone's web interface, to prepend a "1" to a dialed number, you would enter 10-x use the 1$1. Boot Server Configuration - follow these directions:Īs part of adding your Yealink phone to the boot server, the field called "Custom Digit Map" is where the following digit map/dial plan would be added. For assistance beyond the below document, you will need to contact the hardware manufacturer/reseller directly. You are welcome to use the following information to enhance your OnSIP Experience, but technical support is not available from OnSIP for this feature. This is not a standard feature of the OnSIP Hosted PBX. *** This article is for informational purposes only. ![]()
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