Hi Tony,

I've actually established a PPTP vpn (a basic one) from site A to B, and that was the behavior.

I will try again tomorrow. On a separate note, do you have any feedbacks to the polycom 330 templates under */etc/sipxpbx/polycom/mac-address.d *for 2.0 and 3.1; as per the email below?

sip2.2.2 with ROM 4.0 on sipx 3.8 was working normally. After yum 3.10.3 update i've updated the sip3.1.2 and tom 4.1.2 then downgraded to 3.1.1 with 4.1.1 rom and still hanged. the last attempt was to downgrade to 3.1.0 and 4.1.0

when the freeze continued, i switched to good old 2.2.2 with rom 4.0 but suspicious with the polycom 330 template files under
*/etc/sipxpbx/polycom/mac-address.d

*Anything you can shed some light on?
cheers!

Tony Graziano wrote:
Cuney,

I don't think just 'opening" ports at either end would work. If you are passing out over the open internet and have a firewall, it either needs to be sip aware and configured properly, or you need to be able to pass the traffic over a vpn and be able to resolve the DNS SRV records accordingly.

Tony

>>> Cuneyt M 02/04/09 1:27 PM >>> Hi Michael,

I actually follow the tutorial (and verified SRV records resolved at each end) and setup the two end-points. Used the full domain in the gateway address field (also tried with IP addresses, using domain name proved to be working v.s. IP)

What happens is, when i dial from site A and site B phone rings then hangs up. When i dial again from site A to B, B rings and I manage to answer. After about 12 seconds it hangs up.

Ports 5060 and 5090 (both tcp+udp) are opened on site A and B's router.

Does this behavior ring any bells or a pattern?

Offtopic: I've had continuous freezing after i update 3.8 to 3.10.3 and firmware 2.2.2 to 3.1.1 (rom 4.0 to 4.1.1, even tried 4.1.0). I have tried the combinations that are suggested in the list but the freezing is still happening.

As i suspected of a possible corruption in xml configuration, i've exported the user+phones, deleted everything under TFTPROOT folder and import back the user+phones and the Polycom 330 files and deployed the configs after editing Phones under a single Phone Group (i.e. set the date time to 24 hour and disable most of the Features as it doesnt apply to 330).

Now i'm still running 3.10.3 and switched the firmware back to 2.2.2 and ROM 4.0

What i suspect is, the polycom template files under */etc/sipxpbx/polycom/mac-address.d *now changed to 3.1.(1) config XMLs. And if it did i don't know how to reverse it back to 2.0 XMLs.

Do you have any knowledge on this or how to verify and/or switch back to correct XML templates (i am assuming these files under mac-address.d updated with the Yum sipx 3.10.3 update...)

Any help would be greatly appreciated.

Cheers.
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Today's Topics:

   1. Re: multi-site sipXecs deployment (Picher, Michael)
------------------------------------------------------------------------

Subject:
Re: [sipx-users] multi-site sipXecs deployment
From:
"Picher, Michael" <[email protected]>
Date:
Wed, 4 Feb 2009 12:01:44 -0500
To:
"Robert Joly" <[email protected]>, <[email protected]>

To:
"Robert Joly" <[email protected]>, <[email protected]>


http://sipx-wiki..calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs <http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs>

The big thing is to get the two dns domains so they can resolve each other’s addresses…

Mike

*From:* [email protected] [mailto:[email protected]] *On Behalf Of *Robert Joly
*Sent:* Wednesday, February 04, 2009 9:34 AM
*To:* [email protected]
*Subject:* [sipx-users] multi-site sipXecs deployment

Hi,
I have a two-site setup with a sipXecs at each site and I want to connect the two sites together so that a user at a given site can call a user at the other site. I was looking through the sipXecs wiki to figure out the best practices for deploying such a configuration but didn't find anything. If someone has a pointer to such information, could you please send it to me?

Thank you.
bob

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