[VoiceOps] Cisco 7941 SIP

Thu Oct 8 08:43:52 EDT 2015

The “Tradeup2Polycom” landing page is returning a page not found error so you may be right. There are however third party companies that will still buy the 79X1 phones for refurb and parts so you could still do the trade-in as a two-step process.

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Rob Dawson
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From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Wayne Wenthin
Sent: Wednesday, October 07, 2015 12:20 PM
To: Aryn Nakaoka 808.356.2901
Cc: voiceops at voiceops.org
Subject: Re: [VoiceOps] Cisco 7941 SIP

They used to.   I think that program has ended.

On Wed, Oct 7, 2015 at 9:12 AM, Aryn Nakaoka 808.356.2901 <anakaoka at trinet-hi.com<mailto:anakaoka at trinet-hi.com>> wrote:
Maybe Polycom will give you a discount for replacing a Cisco.

Aryn H. K. Nakaoka
anakaoka at trinet-hi.com<mailto:anakaoka at trinet-hi.com>

Direct: 808.356.2901<tel:808.356.2901>
Fax : 808.356.2919<tel:808.356.2919>

Tri-net Solutions
733 Bishop St. #1170
Honolulu, HI 96813
http://www.trinet-hi.com

https://twitter.com/AlohaTone

Aloha Tone PBX <https://www.youtube.com/watch?v=96YWPY9wCeU>  https://www.youtube.com/watch?v=96YWPY9wCeU<http://youtu.be/27v2wbnFIDs>
Aloha Tone (HA) High Availability<http://youtu.be/rJsr4k0RBH8> http://youtu.be/rJsr4k0RBH8

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On Wed, Oct 7, 2015 at 6:11 AM, Aryn Nakaoka 808.356.2901<tel:808.356.2901> <anakaoka at trinet-hi.com<mailto:anakaoka at trinet-hi.com>> wrote:
You can get Polycom phones VVX 101/201 probably for less than the labor hours you will lose on support, not to mention the marketing on that opportunity. I'm sure management would highly consider it.

Aryn H. K. Nakaoka
anakaoka at trinet-hi.com<mailto:anakaoka at trinet-hi.com>

Direct: 808.356.2901<tel:808.356.2901>
Fax : 808.356.2919<tel:808.356.2919>

Tri-net Solutions
733 Bishop St. #1170
Honolulu, HI 96813
http://www.trinet-hi.com

https://twitter.com/AlohaTone

Aloha Tone PBX <https://www.youtube.com/watch?v=96YWPY9wCeU>  https://www.youtube.com/watch?v=96YWPY9wCeU<http://youtu.be/27v2wbnFIDs>
Aloha Tone (HA) High Availability<http://youtu.be/rJsr4k0RBH8> http://youtu.be/rJsr4k0RBH8

CONFIDENTIALITY NOTICE:  The information contained in this email and any attachments may be privileged, confidential and protected from disclosure.  Any disclosure, distribution or copying of this email or any attachments by persons or entities other than the intended recipient is prohibited. If you have received this email in error, please notify the sender immediately by replying to the message and deleting this email and any attachments from your system. Thank you for your cooperation.

On Tue, Oct 6, 2015 at 5:03 PM, Peter E <peeip989 at gmail.com<mailto:peeip989 at gmail.com>> wrote:
You're preaching to the choir, Mark. As a company, for BYOD, we take a stance of, we'll supply the SIP credentials but we won't support the device. But anyone in an operations role knows what that really means -- do whatever it takes to get them working and happy.

I'll share your comments with those that believe the opposite about BYOD and scale. It will make for an interesting debate.

On Oct 6, 2015, at 22:52, Mark Lindsey <lindsey at e-c-group.com<mailto:lindsey at e-c-group.com>> wrote:
1. In Hosted PBX, accommodating new, non-productized devices that the customer just has to keep is the price you pay to enjoy slow growth (because the engineering effort for the customer is immense), poor reliability (because you can test much less), and an unsupportable customer deployments (because the support team isn't equipped to support this "product").

2. In Hosted PBX, the demarc is the audible voice on the speaker and the input to the microphone. Supporting random devices the customer brings you makes it impossible for you to fulfill your end of the bargain: make this voice stuff work every time for every call.

3. The best thing to do with a customer's old device is trade in credit then liquidate.

4. Cisco 79xx SIP has gone back and forth on symmetric sip signaling over the past few decades. But generally, when nat is involved, the sip phone has to do symmetric sip ports -- I.e., it must use the same port numbers for both sending sip and receiving sip. (And when carrier SBCs are involved, it needs to use the same port number for all sip transactions, not just those related to direct call control).

But I remember Cisco 79xx configs having a "nat_enable" or similar flag that actually enable the symmetric sip.

mailto:mark at ecg.co
tel:+1-229-316-0013 http://ecg.co/lindsey

On Oct 6, 2015, at 17:10, Pete E <peeip989 at gmail.com<mailto:peeip989 at gmail.com>> wrote:
Greetings Voice Operators,

We have an interesting (code word for annoying) challenge that we've never dealt with before, probably because we don't do much with Cisco phones. We have a new customer coming on who wants to keep their very old Cisco 7941 phones. They have a few offices and the phones work as expected behind an Edgemarc. However, they also have 100+ home users, and that's where the issue comes in.

Apparently Cisco introduced a security "feature" where they create the session using a random high numbered port (e.g. 49123) but in the Via header, they say to respond to private IP, port 5060. So when the SBC sees the private address it assumes it is being NAT'd through a firewall and replies back to public IP, port 49123. What we're seeing is that the home router passes the response back to private IP, port 49123, which the phone doesn't accept (because it wants it on 5060) and the REGISTER fails.

As you know most home routers are poor at handling ALG (and we've tested and found they are equally bad at handling this scenario). We (and the customer) don't want to troubleshoot 100+ individual home routers.

We haven't found a way to turn off this really awesome "feature" so we're trying to find other solutions. Anyone been through this and have any suggestions?

Thanks,
Pete
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--
Wayne Wenthin
Wide Area Network Administrator
Cascade Technology Alliance (CTA North - Multnomah ESD)
Ph: 503.257.1562
wayne.wenthin at cascadetech.org<mailto:wayne.wenthin at cascadetech.org>
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