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5.0 3
- Thread starter IPhello
- Start date
- Status
any idea on what 5.0 license key will cost?
Isaac Braca
Avaya ACA - IP Office
CTO / ICCS & Co., LLC.
ICCS, Your Premier IT & Telecom Partner, is a New York City Based Avaya SMB Expert Business Partner and IT Consulting Firm.
Visit and Post on my Blog:
Isaac Braca
Avaya ACA - IP Office
CTO / ICCS & Co., LLC.
ICCS, Your Premier IT & Telecom Partner, is a New York City Based Avaya SMB Expert Business Partner and IT Consulting Firm.
Visit and Post on my Blog:
Q3 is the planning but this can be changed
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!
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!
NoelOhashi
Programmer
I have tried using the 9650 IP phone to IPO IP500, version 4.1.15, as one dude had posted here. But no luck. Can some one please show me what txt files should be installed in Manager directory? MIB file? Cause I have tried many txt files, like version 1.1.x 46xxsettings.txt file, version 1.0.x 96xxmib.txt.
I'm desperate!!!
I'm desperate!!!
Noel, i wouldnt bother, you will never be able to upgrade the system, unless they decide to support them in the future.
best ditch the phones now
arsene knows......
everything. apart from the importance of depth in a squad. and what a chequebook looks like......
best ditch the phones now
arsene knows......
everything. apart from the importance of depth in a squad. and what a chequebook looks like......
NoelOhashi
Programmer
On IPO version 4.2.11, the 9650 IP tephone loads the One-X Interface, it shows Extension 1019, which I had previously created, you are able to level the ringing sound, but then it constantly keeps trying to connect to the call server, with no success...
But the good news is I've contacted TheTelecomSpot Store yestereday and one sales woman has told me that the next IPO Release (5.0) will support 9600 IP telephone Series. That's a relief for me...
But the good news is I've contacted TheTelecomSpot Store yestereday and one sales woman has told me that the next IPO Release (5.0) will support 9600 IP telephone Series. That's a relief for me...
I had the 9620 working on 4.1.15 on my 406V2, it might be that it doesn't work on the IP500 but like Tom said ditch the phones or put them into storage for now until you can use them or sell them on e-bay. Let me know when you do that as I am sitting here on an S8500 and would love to have one.
Joe W.
FHandw., ACS
If you can't be good, be good at it!
Joe W.
FHandw., ACS
If you can't be good, be good at it!
NoelOhashi
Programmer
Hahahah. You guys are amazing.... How expensive is it to migrate to CM? Im considering moving to it. IPO IP500 SIP trunk is only stable on version 4.2.11. And in any of its released versions. Out of Band DTMF still is not enabled. So our AutoAttendant menus are useless and trouble makers. Besides I love the 9650 IP phone!!!
amriddle01
Programmer
clarificatrion fo GoBurko
Field Trial Timescales
• Field trial software will be available 20th April 2009
• Field trial installations to be completed by 4th May 2009
• Intermediate builds of software released as required
• GA candidate of Field Trial software available 6th July 2009
• Upgrades of trial sites to GA build to be completed by 13th July 2009
Target GA Release Date 3rd August 2009
ACS - IP Office Implement
"I'm just off to Hartlepool to buy some exploding trousers
Field Trial Timescales
• Field trial software will be available 20th April 2009
• Field trial installations to be completed by 4th May 2009
• Intermediate builds of software released as required
• GA candidate of Field Trial software available 6th July 2009
• Upgrades of trial sites to GA build to be completed by 13th July 2009
Target GA Release Date 3rd August 2009
ACS - IP Office Implement
"I'm just off to Hartlepool to buy some exploding trousers
@NoelOhashi
The IPO supports RFC2833 for DTMF. The Out of Band DTMF using INFO you are referring to as "not enabled" is rarely used. I've been using SIP trunks exclusively on several switches since it was introduced to IP Office without any issues.
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
The IPO supports RFC2833 for DTMF. The Out of Band DTMF using INFO you are referring to as "not enabled" is rarely used. I've been using SIP trunks exclusively on several switches since it was introduced to IP Office without any issues.
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
Programmer
@kholladay
Then help me understand why lots of my clients are having trouble into using the Autoattendant, cause dtmf digits are generally not recognized. Not to mention the fax transmission rarelly works. And I'm using GVT Vono SIP trunks. GVT is top of the fields when it comes to SIP trunk here in Brazil. SIP trunk configured to MU Law, switch and line to ALaw.
Then help me understand why lots of my clients are having trouble into using the Autoattendant, cause dtmf digits are generally not recognized. Not to mention the fax transmission rarelly works. And I'm using GVT Vono SIP trunks. GVT is top of the fields when it comes to SIP trunk here in Brazil. SIP trunk configured to MU Law, switch and line to ALaw.
Do you have RFC2833 enabled on the provider end? In your INVITE you should see "a=fmtp:101 0-15" coming from the IP Office. This tells the provider that the IP Office supports DTMF events 0-9, *, # and A-D.
The provider should send back an fmtp message with their supported DTMF events. Here is an example:
IPO to Provider:
835437366mS SIP Trunk: 20:Tx
INVITE sip:1xxxxxxxxxx@did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport;branch=z9hG4bKee09556c185a0975b244acf5d9e57551
From: "KyleHolladay" <sip:xxxxxxxxxx@mysipline.net>;tag=69bd8f80bfd83b8f
To: <sip:1xxxxxxxxxx@mysipline.net>
Call-ID: 8c4d52ae1c30ef975c1ddd9808f48269@xx.xx.xx.xx
CSeq: 1682787794 INVITE
Contact: "KyleHolladay" <sip
rimeServicesLLC@xx.xx.xx.xx:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 277
v=0
o=UserA 1962482231 1291585475 IN IP4 71.39.23.171
s=Session SDP
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 49154 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Provider to IPO
835442048mS SIP Rx: UDP xx.xx.xx.xx:5060 -> xx.xx.xx.xx:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKee09556c185a0975b244acf5d9e57551;received=xx.xx.xx.xx;rport=5060
From: "KyleHolladay" <sip:xxxxxxxxxx@mysipline.com>;tag=69bd8f80bfd83b8f
To: <sip:13033502352@did.voip.les.net>;tag=as0768fb8e
Call-ID: 8c4d52ae1c30ef975c1ddd9808f48269@xx.xx.xx.xx
CSeq: 1682787794 INVITE
User-Agent: SIP.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1xxxxxxxxxx@xx.xx.xx.xx>
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 16383 16383 IN IP4 xx.xx.xx.xx
s=session
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 16498 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp
ff - - - -
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
The provider should send back an fmtp message with their supported DTMF events. Here is an example:
IPO to Provider:
835437366mS SIP Trunk: 20:Tx
INVITE sip:1xxxxxxxxxx@did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport;branch=z9hG4bKee09556c185a0975b244acf5d9e57551
From: "KyleHolladay" <sip:xxxxxxxxxx@mysipline.net>;tag=69bd8f80bfd83b8f
To: <sip:1xxxxxxxxxx@mysipline.net>
Call-ID: 8c4d52ae1c30ef975c1ddd9808f48269@xx.xx.xx.xx
CSeq: 1682787794 INVITE
Contact: "KyleHolladay" <sip
rimeServicesLLC@xx.xx.xx.xx:5060;transport=udp>Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 277
v=0
o=UserA 1962482231 1291585475 IN IP4 71.39.23.171
s=Session SDP
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 49154 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Provider to IPO
835442048mS SIP Rx: UDP xx.xx.xx.xx:5060 -> xx.xx.xx.xx:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKee09556c185a0975b244acf5d9e57551;received=xx.xx.xx.xx;rport=5060
From: "KyleHolladay" <sip:xxxxxxxxxx@mysipline.com>;tag=69bd8f80bfd83b8f
To: <sip:13033502352@did.voip.les.net>;tag=as0768fb8e
Call-ID: 8c4d52ae1c30ef975c1ddd9808f48269@xx.xx.xx.xx
CSeq: 1682787794 INVITE
User-Agent: SIP.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1xxxxxxxxxx@xx.xx.xx.xx>
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 16383 16383 IN IP4 xx.xx.xx.xx
s=session
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 16498 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp
ff - - - -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
Programmer
Heres what I monitored, while testing our SIP trunk:
3011167mS CMMap: PCG::UnmapBChan pcp[41]b1r0 cp_b f5a1c944 other_cp_b f5a21368
3011167mS CMMap: a=6.17 b=21.9 M0
3011167mS CMMap: PCG::UnmapBChan pcp[239]b0r1 cp_b 0 other_cp_b 0
3011167mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
3011167mS H323Evt: RTP(END): 189.120.170.101/49154 201.86.87.2/11104 CODEC 5 PKTSZ=160 RFC2833=on AGE=39757 SENT 1980 RECV 1976 RTdelay=0 jitter=0 loss=256 remotejitter=0 remoteloss=0
3011168mS CMMap: a=6.17 b=0.0 Mapper::FreeCodec freed CMRTVocoder resource busy 3, total 12
3011169mS CD: CALLSYNC: cs02
3011310mS RES: Thu 19/2/2009 16:24:52 FreeMem=73436228(1) CMMsg=5 (5) Buff=200 938 999 7416 5 Links=2069
3011310mS RES2: RTEngine=0, CMRTEngine=0, Timer=62, Poll=0, Ready=1, CMReady=0, CMQueue=0, VPNNQueue=0
3016152mS PRN: Destroyed MH f5477e20 parent unknown
3016153mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
3016153mS H323Evt: RTP(END): 189.120.170.101/49152 201.86.87.2/11890 CODEC 6 PKTSZ=20 RFC2833=on AGE=44685 SENT 1976 RECV 1973 RTdelay=0 jitter=0 loss=256 rem
3011167mS CMMap: PCG::UnmapBChan pcp[41]b1r0 cp_b f5a1c944 other_cp_b f5a21368
3011167mS CMMap: a=6.17 b=21.9 M0
3011167mS CMMap: PCG::UnmapBChan pcp[239]b0r1 cp_b 0 other_cp_b 0
3011167mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
3011167mS H323Evt: RTP(END): 189.120.170.101/49154 201.86.87.2/11104 CODEC 5 PKTSZ=160 RFC2833=on AGE=39757 SENT 1980 RECV 1976 RTdelay=0 jitter=0 loss=256 remotejitter=0 remoteloss=0
3011168mS CMMap: a=6.17 b=0.0 Mapper::FreeCodec freed CMRTVocoder resource busy 3, total 12
3011169mS CD: CALLSYNC: cs02
3011310mS RES: Thu 19/2/2009 16:24:52 FreeMem=73436228(1) CMMsg=5 (5) Buff=200 938 999 7416 5 Links=2069
3011310mS RES2: RTEngine=0, CMRTEngine=0, Timer=62, Poll=0, Ready=1, CMReady=0, CMQueue=0, VPNNQueue=0
3016152mS PRN: Destroyed MH f5477e20 parent unknown
3016153mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
3016153mS H323Evt: RTP(END): 189.120.170.101/49152 201.86.87.2/11890 CODEC 6 PKTSZ=20 RFC2833=on AGE=44685 SENT 1976 RECV 1973 RTdelay=0 jitter=0 loss=256 rem
Escorthosis
Technical User
To compare you're results with Kholladay you should tick
the 'sip' setting in the filter\trace options from the monitor program.
the 'sip' setting in the filter\trace options from the monitor program.
NoelOhashi
Programmer
@ Escorthosis
Many thanks. I will do it again tomorrow at my office. You re very kind. I hope I can solve this issue soon, or my clients will keep on pushing my patient to the limits.
Many thanks. I will do it again tomorrow at my office. You re very kind. I hope I can solve this issue soon, or my clients will keep on pushing my patient to the limits.
@NoelOhashi,
Are you using VMPro?
If so check if "Allow prompts to be interrupted by Tones" is ticked in the Menu "Entry Prompts" tab.
Greetzzz...Bas
___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
Are you using VMPro?
If so check if "Allow prompts to be interrupted by Tones" is ticked in the Menu "Entry Prompts" tab.
Greetzzz...Bas
___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
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