5.0 3

[kholladay]

We have used Kentrox without issue. We are now using Adtran.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[NoelOhashi]

@ Kholladay

Kentrox ? Really? And why dont you use Kentrox anymore? How many SIP calls can it handle?

 

[kholladay]

Kentrox discontinued the Q series routers. Adtran has been a good replacement. We use the 3430 and in the case where a few PoE ports may be needed for a remote office we use the 3448 which has (with the PoE power supply) 8ea PoE Ethernet ports.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[tanderson3733]

QoS is a moot point on the public Internet. No provider will allow you to prioritize your traffic over other customer's for any reason, and all means of tagging packets or otherwise are dropped or stripped.

 

[kholladay]

Not at all tanderson3733. Traffic over the network is only one part of the problem. You can significantly improve your voice quality by prioritizing traffic TO the network (i.e. at your gateway/router). If I have already queued my voice traffic when it hits the router then I am guaranteed issues. If I can insure that voice traffic hitting the Internet goes out without delay/queuing then I am already a step in the right direction. I can hear a significant difference in my daily voice traffic when I use a Kentrox or Adtran where my voice traffic is prioritized vs when I use my SonicWall where it is not. On the SonicWall if I send a large email attachment or I start up uTorrent, thus taking my bandwidth to near capacity, I hear the difference. With the Adtran the data traffic is queued to allow the voice priority, I hear little to no difference. Goes back to the old programming phrase, garbage in garbage out. You dump poor voice quality into your Internet connection and you're sure to get poor voice quality on the other end. Dump good voice quality in and your far more likely to get acceptable to good quality on the other end.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[NoelOhashi]

I have just noticed that everytime one SIP connection is established to any of the users, the called party hears distortions when the caller is asked to press any key of his phone. What could be causing these distortions? Avaya IPO or Linksys router?

 

[amriddle01]

Why are they asking them to press keys while they are talking to them? You would normally hear DTMF when they do this but I don't see why they would want too.

ACS - IP Office Implement

"I'm just off to Hartlepool to buy some exploding trousers

 

[Westi]

SIP line or SIP extension seeing that you post it in the R5.0 thread
silence suppression on? if you have R5 then is there an option for inband or out of band dtmf?
otherwise try a new thread if it is a SIP line as this one is the never ending thread already


Joe W.

FHandw., ACS

If you can't be good, be good at it!

 

[tlpeter]

R5.0 is not out
Fieldtrials are not started yet

ACA - Implement IP Office
ACS - Implement IP Office
ACA - Implement IP Telephony
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[Westi]

in that case it can't be a SIP extension I guess

I will try to get in to the field trials then too, maybe I get lucky

Joe W.

FHandw., ACS

If you can't be good, be good at it!

 

[NoelOhashi]

@ amriddle01
Pressing keys while we're talking to callers is just to prove that DTMF tones come distorted on Avaya IPO IP500 SIP trunking, explaining why DTMF isnt getting recognised on IPO IP500 SIP trunking, in my scenario. I use one Linksys BEFSR41 router to connect the SIP trunk to the IPO. Has anyone ever tried using Edgewater Networks EM 4500 WAN routers? Do they work better on a Broadband connection?

 

[jamie77]

We used to run SIP trunks in an IP500 using Zyxel routers without issue (on private DSL) and are now running them through Draytek Vigor (over shared internet DSL) also without issue.

Both setups on Voiceflex SIP Trunks (UK).

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office

Football is not a matter of life and death-It is far more important!!!!

 

[NoelOhashi]

@ jamie77

Hi Jamie, I just got one DrayTek 2910. Many thnaks, but which DiffServ CodePoint would u suggest me to use? AF Class 2 High Drop? I need some help with it.

 

[NoelOhashi]

@ EveryOne
Announcing IP Office Release 5

It's all about greater business continuity, by introducing Resilient Small Community Network. To have a second site be a backup for a main or another site and to have voicemail (pro) re-register to a backup site is really approaching a new status for reliability. This could only be accomplished though, with IP endpoints (phones). Digital or analog phones connected to a failed system would not have this ability. With this enhancement, hunt groups and users will be known on both systems and most useable features are transparent There will not be idle hardware in a design implemented with the redundant SCM. Another great enhancement to SCN is the expansion to 32 sites, doubling the systems that can be connected. There's also word that release 5 will support more phones, extending options for users with needs for color and enterprise phones. There may not seem like much that's new with release 5, but it is a lot when considering how this will affect the decision to design and implement a resilient communications platform. Targeted availability is sometime in late summer 2009.

Font:

http://www.merchantcircle.com/blogs...tinuity-Announcing-IP-Office-Release-5/205776

 

[NoelOhashi]

According to one important staff in Avaya, yes, sometime near Sept 2009, version 5.x will support 96xx phones. After all, we need more color to our desk and better voice transmission on SIP trunks.

 

[NoelOhashi]

@Everybody
Hi everybody. Ive upgraded our Avaya IP500 firmware to 5.0, but still I face DTMF issues on our SIP trunk incoming calls. Since I still had issues when using one broadband router (Draytek 2910) I decided to run the SIP trunk directly from our broadband internet cable modem. I found no issues on outbound calls, but still I cannot have DTMF detection when I receive inbound calls from my Nokia E61 GSM cell phone or my Panasonic cordless phone (Avaya IPO SOE analog extn attached to one analog phone line) or my Avaya 4624 IP phone, attached to one analog phone line.

Does anyone have any similar experience?

 

[Bas1234]

@NoelOhashi,

Default Avaya uses Out-of-Band (RFC 2833).
Check whit the provider if they can change it, if not then ask what they use and change it in the IPO.

Line > SIP Line > Voip tab >DTMF Support.

Also posting a trace will help.

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[NoelOhashi]

@ Everyone

The issue was caused by the SIP line provider.... I just moved to another SIP line provider and everything works smoothly. I love Avaya IPO ...

 

[Westi]

NoelOhashi
we all do, most of the time
SIP providers are a funny thing sometimes, you should assume that the SIP standard will be followed but sometimes that seems just wishful thinking.

Joe W.

FHandw., ACS

If you can't be good, be good at it!