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Combination bug EdgeMarc/Polycom?
by owen＠impulse.net Jan. 10, 2011

Jan. 10, 2011

Hello Voice-Ops, I have a frustrating situation where a random 1% of PolyCom phones (Mix, but all running 3.1.1) behind EdgeMarcs (VRRP 9.1.9) configuration are losing registrations overnight after a failed boot server check-in. (FTP) Phone will be URL call disabled until reboot or 12 hour+ wait. Inbound calls still work, outbound calls fail. Registration from a packet level is normal. Scenario is complex, but repeatable at multiple client sites. (so not firewall/lan) wireshark, tech support, release codes, ftp logs, and boot logs are all unhelpful/empty, and the firmware code (s) we're running can't be changed without proof, which at this point can only be generated by a fun weekend spent rebooting a cluster of phones over and over again, which for an annoying bug seems excessive. Bit of a rim-rock. Anyone seen anything like this? Cheers, Owen R. ____ Senior Network Engineer Impulse Advanced Communications 805-884-6332 owen at impulse.net

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Terminating LD Traffic
by clec＠cygnustel.com Jan. 9, 2011

Jan. 9, 2011

>From my understanding if you want to terminate Long Distance traffic to the LECs exchanges you have to have end office trunking. Being that end office trunks are special access the only affordable way to do this is to be collocated in the exchange or have 3rd party trunking into the exchange. I would like to start looking into this as we are having to collocate in an increasing number of exchanges and I would like to try to recoup some money by terminations if possible. Does anyone else do this or have any advice on the matter? Thanks guys! -Eric

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Terminating LD Traffic
by naushit＠yahoo.com Jan. 9, 2011

Jan. 9, 2011

Eric, You did not mentioned LATA or Area, I assume you are looking for nation wide connectivity? We are New York based CLEC, and we have plenty of extra bandwidth to ILEC End offfice and Tandem office. We would be happy to share bandwidth at cost and collocate your equipment in our data center in most affordable way. We are also interested in trading Chicago DIDs ( or any other area DIDs too) for NYC DIDs . Please contact me off the list. -Naush. From: Eric Hiller <clec at cygnustel.com> To: voiceops at voiceops.org Cc: Sent: Saturday, January 8, 2011 7:25:39 PM Subject: [VoiceOps] Terminating LD Traffic >From my understanding if you want to terminate Long Distance traffic to the LECs exchanges you have to have end office trunking. Being that end office trunks are special access the only affordable way to do this is to be collocated in the exchange or have 3rd party trunking into the exchange. I would like to start looking into this as we are having to collocate in an increasing number of exchanges and I would like to try to recoup some money by terminations if possible. Does anyone else do this or have any advice on the matter? Thanks guys! -Eric _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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"Ring Down" Phone
by robert.dawson＠mindshift.com Jan. 7, 2011

Jan. 7, 2011

I am sure I have seen these before but I am not turning anything up . . . basically looking for an endpoint with no keypad that is programmed to dial a specified number when it goes off hook - just like a door box. Anyone know of anything offhand? Thanks, Rob

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Porting numbers from Centurytel
by mylists＠battleop.com Jan. 4, 2011

Jan. 4, 2011

Does anyone know of a SIP provider that can port numbers from Centurylink ILec markets across the country? We run into this often where a customer has remote locations that are in Centurylink territory and we can't port their numbers back to our network. Most of these markets have small Centurylink footprints or they are somewhat rural. Richey

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Polycom Ghost Ringing
by clec＠cygnustel.com Jan. 3, 2011

Jan. 3, 2011

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Call setup on these calls is all normal: invite from pbx to phone 100 trying from phone to pbx 180 ringing from phone to pbx 200 ok from phone to pbx ack from pbx to phone <start rtp> I have listened to the RTP and there is absolutely no ringing! At this point I am entirely out of ideas! Does anyone have anything to suggest or has possibly encountered this before? Thanks for any ideas or help! -Eric

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Is call forwarding in Polycom phones fundamentally broken?
by jdupuy-list＠socket.net Jan. 3, 2011

Jan. 3, 2011

I've encountered this problem in the past, many years ago, when working with Asterisk farms. I'm encountering it again now working with the Broadsoft platform: Polycom Soundpoint IP phones can do call forwarding in two ways: client-side and server-side. In client-side mode, the phone itself does the forwarding. If the phone is unreachable for any reason, the call forwarding never actually happens. That is because the call server doesn't know to forward the call and the phone isn't reachable to do the forwarding. By default, the call usually goes to VM, which is not what the user wanted. So, it works most of the time, but not predictably. So, it is broken by design. (Plus with the half-broken NAT support on the Polycoms, any forwarded calls needlessly burn two audio streams because the call is not likely to transfer off the handset. But, I'll refrain on Polycom NAT commentary for the moment...) So, we go with server-side... In server-side mode, the phone refuses to show that the phone is forwarded anywhere on the screen. It is not uncommon for the user to "wonder why they aren't getting calls" if they forget to cancel the forwarding. So, it is also broken by design. (Albeit, not as badly as client-side, since at least it will always function.) This does not appear to be a bug, since it is well documented by Polycom. The only work-around I've come up with is writing a customized hack that modifies the idle screen, but that is far less than ideal. :) In general, I really like Polycom phones, but may have no choice but to switch to cisco/aastra/etc. But, before I do, has anyone come up with a better work-around? John

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Mexico DID's
by sil＠infiltrated.net Dec. 30, 2010

Dec. 30, 2010

Hey all, while not related to operational issues, still VoIP related! Lest I forget, happy new years to everyone here, so far I seem to enjoy and appreciate being able to learn from the others on the list and share when I can. Anyhow, down to "bidness," can anyone recommend a company providing DID's in Mexico, specifically the central region (Queretaro). I'm aware that GLBX has a presence there, but does anyone know of an alternative/failover? -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E

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Dynamic T1 Faxing over IP
by ciscoplumber＠gmail.com Dec. 28, 2010

Dec. 28, 2010

We are experiencing a fair amount of heart burn attempting to support faxing over our Dynamic T1 product. Some customers it works great and others have frequent problems. We currently support dynamic T1's using both Adtran TA900 series and Cisco IAD2400 services routers. The customers fax machines will connect to their PBX that has a PRI to an IAD/TA. We have the IAD/TA configured for SIP signaling to an Acme SBC selecting T.38 as the preferred codec with fallback to g.711 pass-through. We can force a slower speed, disable error correction, disabled VAD but in the end still have very inconsistent results. From a network perspective the QoS is matching and queuing correctly, the circuits are clean and the IP SLA gives us no reason to believe they are having any network drops or jitter. I was wondering if others who have experience with faxing over IP could share what they have learned works best or anything else that may be helpful. Any help is greatly appreciated. Thanks, Jeff

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AOC for SIP
by Kevork.Belian＠spidernet.com Dec. 28, 2010

Dec. 28, 2010

Hi folks, Does anybody know about AOC of charge implementation on SIP Proxy. What I am looking for is a SIP server that can be configured as Advice of Charge designation point and send AOC messages via SIP INFO. Astersik has this capability with Sirrix PCI cards. But I am looking for something without any card dependency. thanks Kevork Belian PrimeTel Senior Network Engineer The Maritime Centre, 141 Omonoia Avenue, CY-3506, Limassol, Cyprus e-mail: kevork at francoudi.com www.thunderworx.com<outbind://192/www.thunderworx.com> tel.: (+357) 25867136 fax: (+357) 25568131

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