PSTN Gateway problem
I've been having a go with making calls to my home through the PSTN gateway number (02) 8307 8846.
I have a SPA-3000 at home which registers to VoXaLot. Outgoing calls work fine. When the PSTN gateway answers I enter *010999999 at the prompt (*010 for VoXaLot, 999999 obviously replaced by my VoXaLot account number). The phone rings at home but when answered neither party can hear anything. Any pointers to help me troubleshoot this one? |
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Usually the fault of the firewall/NAT blocking traffic. What are your STUN settings in the SPA-3000? How about if someone calls your Voxalot address directly by a SIP URI call via something like the Gizmo Project softphone--does two way audio work there?
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It would be:
sip:123456@voxalot.com |
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Normal calls work over my firewall - would I need different settings in the firewall for SIP calls? STUN Settings on the SPA SIP tab are: NAT Support Parameters: no Handle VIA received: no Handle VIA rport: no Insert VIA received: no Insert VIA rport: no Substitute VIA Addr: no Send Resp To Src Port: no STUN Enable: no STUN Test Enable: no STUN Server: EXT IP: EXT RTP Port Min: NAT Keep Alive Intvl: 15 Anything there that looks incorrect for SIP calls? |
What if you tried these settings?
* Handle VIA received: no * Handle VIA rport: no * Insert VIA received: no * Insert VIA rport: no * Substitute VIA Addr: yes * Send Resp To Src Port: yes * STUN Enable: yes * STUN Test Enable: no * STUN Server: stun.voxalot.com.au:3478 o NOTE: You can replace the above STUN server with any STUN server you like... * EXT IP: o NOTE: Leave this setting blank, STUN will figure this out for you... * EXT RTP Port Min: o NOTE: Normally you can leave this blank, but you can set this if you have a specific need * NAT Keep Alive Intvl: 45 o NOTE: Use an value SHORTER than the "timeout" value in your router. In the VOIP tab, set "NAT Mapping Enable: yes" set "NAT Keep Alive Enable: yes" |
Philled, I had this problem. I think that it was because my ATA was registed with voxalot.com.au but I was using *010. When I used *061 it worked OK. I only tested a couple of times but I think that this was the problem.
MG. |
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udp 5060:5061 # Initiate call udp 16384:16482 # Transmit voice udp 35000:45000 # For SIPME Regards, Phill |
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