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Full Version: Sip Phone Cuts Off For Outgoing Calls After 20 Sec
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This is a question for the Geeks FC.Net. Currently have a Cisco IP Phone at location 1 (Home) connected to our * box at location 2. Our phones are pre-programmed to a specific public IP Address/Host name that is publically reachable (non ipsec, non vpn tunneling) for provisioning purposes.

The incoming cals work just fine and reach my extention properly, its only outgoing that cut off after exactly 20 seconds. Now before some of you jump on me for firewall and port forwarding, i have turned off firewall completely, and put the ip phones ip on DMZ and still nothing. which leads me to believe its a port issue on the * box?

any advise?
who's your internet provider at home (location 1)?

unfortunately, some internet providers don't like you making outgoing connections on services you didn't buy from them... so they do a little trick where they renew your IP address on you... which effectively breaks some connections that are trying to get back to you.

I've noticed this with **** through my Bell DSL... and I've had to go through some of the software configuration to re-establish the connection when the IP changes.

So what I'd do, is simply get your exterior facing IP address before the call, then after it drops and see if they differ. If so, you'll need to find some configuration around it, otherwise, you may need to get tech support from your provider to sort it out.

And if they're like Rogers, they'll ask you to upgrade to a VPN-compliant business package (though, most VPNs can work around it).

rogers at both location, its not an issue with the ip being changed, we have a static host name at location #1 where our ip phones are provisioned to look for this server and establish a connection with. something tells me its a setting on the * box.
do u think i should try an SS7 trace on the terminating number to see if its an issue with the trunk/route?
Do you have access to the SS7 gateway to do a trace? That's usually where it's done from (IIRC).

And, I'm not the biggest teleco expert around, but is the Cisco IP phone SS7 over IP, or VOIP?

Chances are it's something IP-related and not necessarily SS7 related and I'd probably check your IP routing first. If you can determine what/which packets are being lost, that might be more insightful than doing an SS7 trace first.
of course i have access to the ss7 gateway, what cellphone repair jockey does not?

anyways, you may be onto something here dark, it might be a routing issue. I'll look into it more, the ss7 thingy is easier to do anyways, so ill do it first, then i'll try playing with the routing, this is a little more complex cus I hafta move to another location and open/close ports, more time consuming. but i'll keep you posted on if it works.
TTT

Any ideas?
paolo,Apr 3 2010, 10:04 AM Wrote:TTT

Any ideas?
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A string and two cans?

It never cuts out unless you let the string go slack.

:lol:
nice suggestion smart alex, i was hoping for a real responce from a REAL NERD. guess you only just wear the glasses.
paolo,Apr 3 2010, 01:29 PM Wrote:nice suggestion smart alex,  i was hoping for a real responce from a REAL NERD.  guess you only just wear the glasses.
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Sorry, I program robots and automation, not cell phones. Wrong area of nerdspertise. B)
No problem dweeb. Any other nerds out there know?

TTT
Only other thing that I can ask is whether the connection at home is a dedicated one, with nothing else sharing it?

I assume that you've tried your phone with nothing else connected and you still get the same result? :unsure:

NefCanuck
i have a second SIP account with another server located half way around the world, programmed extention 2 in my SIP phone with those credentials, and it works, it looks like the problem is just the port issue with server 1 being behind a firewalled router and not all the proper ports are opened. Heck, the router that is behind is not the greatest it looks like im going to have to replace that.

I have a second question. I have 4 unused extensions so i was thinking if its worth it to have my local pstn line patched into my extention 3 (line key 3) I am thinking about buying a FXO / FXS adaptor, and using it locally only, ie programming it into the residential pstn, to turn it into a gateway, giving it a local internal ip, and programming the third extension of the ip phone with it that way i can get line key 1 will be my telus sip trunk (etobicoke), line 2 from sweeden sip account, and line 3 will be my local pstn. all on the same phone. it shouldn't be really that dificult, no?