Sip Trunks

[PRNDL2]

Installed IPOffice 500v2 rel 8.1 w/ 8 cablevision sip trunks. Outgoing calling has never been a problem, works every time....incoming calls have been a problem. I can use an extension and dial the main number and it rings in correctly every time.....while doing this to test all 8 channels I also used 2 cell phones and they also rang in correctly however if the channels are idle for a period of time and we try calling in via a cell phone only (not an extension) the call does not ring in and is forwarded to a cell phone (setup by the carrier). It seems the sip trunks lose registration when idle but if I dial the main number via an IPO ext the call always rings in correctly.

Any ideas???


Also what setting allows the company's main number to be outpulsed for outbound caller id....now called parties see unavailable.

Thanks in advance for any help.

Jerry

 

[amriddle01]

Firewall issue I'd say, when you call out it opens port 5060 up and that will allow traffic to return for a brief period afterwards, after that time unless you have allowed port 5060 on the firewall it will block traffic originated externally as it's seen as unsoliciited/a hack

 

[PRNDL2]

Installed IPOffice 500v2 rel 8.1 w/ 8 cablevision sip trunks. Outgoing calling has never been a problem, works every time....incoming calls have been a problem. I can use an extension and dial the main number and it rings in correctly every time.....while doing this to test all 8 channels I also used 2 cell phones and they also rang in correctly however if the channels are idle for a period of time and we try calling in via a cell phone only (not an extension) the call does not ring in and is forwarded to a cell phone (setup by the carrier). It seems the sip trunks lose registration when idle but if I dial the main number via an IPO ext the call always rings in correctly.

On the Sip tab of the Sip trunk in manager, does anyone think by changing Update Supported from Never to Allow or Auto might resolve my problem of channels losing registration (falling asleep) for inbound calls? The defination of Update Supported is;

UPDATE Supported: Default = Never. Release 8.0+.

The SIP UPDATE method (RFC 3311) allows a client to update parameters of a session (such as the set of media streams and their codecs) but has no impact on the state of a dialog. It is similar to RE-INVITE, but can be sent before the initial INVITE has completed. This allows it to update session parameters within early dialogs. The settings are Never, Allow or Auto. If set to Allow or Auto and the other party supports UPDATE, the IP Office will send UPDATE messages for session refresh.

Also any ideas on how to send the customer's main number for outbound caller id?? Right now called parties see Unavailable.

Thanks,

Jerry

 

[nothingworks101]

not sure if it helps you but this happened to me: ipo rel 8 with a siperator connection from cablevision


cablevision said they are rejecting the call because the ipo is aking for a supported timer? they said they do not require the timer and that is why the call dosnt complete.

they said they support rfc 3261?

I found:

Require: timer shows up in the release 8 that cablevision says is not supported but is missing in release 6.

release 8 trace:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bKb4fe08580dde760f3d1904ca3f162310.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bKgvh21d0028t1dug1g310.1.tow45Igmvu2f3tsw78pQLQ__
Record-Route: <sip:rJfGZdAnTKqj9BEs.4R12785a5065@172.16.1.1;lr>
From: <sip:9177015942@173.251.12.19:5060>;tag=24.170.214.10+1+3bd137+5f4a65b3
To: <sip:2035424708@24.157.52.202>;tag=a6a8db1a84f21c4f
Call-ID: 32855138@24.170.214.10
CSeq: 395320869 INVITE
Contact: "Extn4708" <sip:2035424708@172.16.1.2:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Min-SE: 14400
Require: timer

 

[acollard83]

TO set your outbound callerid, in the SIP tab of each user, put in the number that you want to send out.