ACS - Implement IP Office
ACA - Implement IP Telephony -- ACA - Design 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!
SHUT UP TALKING ABOUT YOUR OTHER HALVES !!!
You'll all ruin the public's perception of us as socially inept geeks with no social skills and fantasies of revenge with guns!!!
You'll all ruin the public's perception of us as socially inept geeks with no social skills and fantasies of revenge with guns!!!"
Geek sheik is in!
its cool to be uncool. We have won.
My missus actually asked me over dinner the other night if there was a chance we could set up a forum for all our wags to go on and moan about how much time we spend on here.
@amriddle01
Hi. Do you use SIP trunk on this Beta Release. If so... Have they enabled the Out of Band DTMF feature on the SIP trunk tab???
Im about to join the Beta Program here.
Many thanks,
Noel Ohashi
Hey Amriddle01 many thanks. So its worth waiting for Avaya IPO Release 5.0 afterall. August is about to come anyway ! Right? And all Avaya IPO users will be able to download the new IPO version, right?
By the way, have you already tested DTMF performance on VMP autoattendants. after installing version 5? Does it work ?
Im going nuts from listening to customers complaints about DTMF distortion.
How bout fax support? How is it working for you? I'm so curious about it.
Not much of a change from R4 to R5 in that RFC2833 was already the default in R4. The only difference there is the option to use INFO which isn't widespread anyway.
T.38 support is fine but you'll not find many ITSP out there that support it and the header information in the INVITE in R5 is so limited that those who do support it don't seem to support the way Avaya's implemented it. Look for it to better support fax over IP between IPO in an SCN and to offer the ability to connect to approved analog gateways for remote fax devices.
Overall I've been satisfied with the R4 implementation of SIP and R5 simply makes it easier to program (primarily from the ability to wildcard URI on the SIP line). We implement, on average, 5 all SIP trunk IP Offices on R4 each month and I haven't had any complaints.
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
@kolladay
Im using a SIP trunk, running Avaya IPO Release 4.2(11), as version 17 makes the inbound calls work for 10 mins only and then, theres just busy tone for incoming callers. Yet on version 4.2(17), the Out of Band DTMF stays greyed out. It isnt possible to change that. This makes incoming DTMF tone distorted and there's no way to work without hearing our customers compains about how "the autoattendant feature doesn't work". Although I love Avaya solutions I am very concerned with the fact that I can't find a way to solve the DTMF issue on the inbound calls when using SIP trunk. I tried everything. Im already using a new router (Dreytek 2910). This router gives our SIP trunk more availability, but still, most GSM inbound calls coming from Nokia or Motorola cells dont work. iPhone works fine. Does release 5.0 have the feature Out Of Band DTMF enabled? I hope so... as you've told us that versions 4.2 already have RFC2833 enabled.
There is nothing wrong in 4.2.17
It is probably your trunk that does not meet the specs
Avaya is very clear which protocols the support
For DTMF your provider needs to support RFC2833
ACS - Implement IP Office
ACA - Implement IP Telephony -- ACA - Design 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!
Out of band DTMF is "INFO" not RFC2833. The two options in R4 are in-band and RFC2833. RFC2833 being the default. If you look at your INVITE you should see:
"“0-15” denotes the DTMF events supported within the “telephone-event” payload (DTMF digits 0 to 15). Instead of sending actual DTMF tones, representations of the DTMF tones are sent (what key pressed and for how long) and the receiving site is expected to reproduce the DTMF tones locally (RFC 2833)."
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
maybe your re-invite is not working (hence the busy signal incoming and probably outgoing OK) with your specific line provider with SIP there is a known issue and a special built 4.2(172202)for it.
I stole that info from another thread.
There isn't anything to set. If the SIP provider offers up RFC2833 the IP Office will accept it. If they don't then it will revert to in-band DTMF. On outbound calls the INVITE will always request RFC2833 (digits 0-15 see table below) and wait for the far end to accept or negotiate in-band.
Hi... Ive just tested anyother SIP provider here in Sao Paulo, and now DTMF works wonders. Even if the caller is a cell phone user. Tomorrow, I will test T38 transmission.
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