CUBE - DTMF Interworking between SIP INFO and RFC2833
Hi all,
After some testing I noticed that when the cube receives dtmf via SIP INFO it doesn't translate them into rfc2833 on the other side. This happens when the incoming INVITE advertises both SIP INFO and RFC2833.
When the incoming INVITE advertises only SIP INFO then it does the translation to RFC2833 and sends the digits via RTP on the outgoing interface.
CUBE supports both RFC2833 and SIP INFO as it advertises both methods in it's sip messages. Is this the expected behaviour ?
IOS is 15.1(3)T1. CUBE is running on a 3945 router.
I have found bug CSCuh65102 which seems to be a match. Can somebody confirm this ?
Hi,
Please check the following bug
https://tools.cisco.com/bugsearch/bug/CSCtj93573/?reffering_site=dumpcr
CUBE not processing DTMF SIP INFO to RFC 2833 upstream to Network
You can try an upgrade to 15.1(3)T4 or higher and check if the issue is resolved.
HTH
Manish
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PdfA1b - Subject Mismatch between Document Info and XMP metadata
As far as I can tell, when generating PdfA-1b files I am completely compliant with the spec. Yet when I try to preflight verify my output I get the following error: "Subject mismatch between Document Info and XMP metadata".
Here is my Document Info:
/Title (Title)
/Author (Author)
/Subject (Subject)
/Keywords (Key Words)
/Creator (Creator)
/Producer (Producer)
/CreationDate (D:00010101000000)
/ModDate (D:00010101000000)
Here is my XMP Stream:
Key Words
Producer
Title
Subject
Author
Creator
0001-01-01T00:00:00.0011111
0001-01-01T00:00:00.0022222
1
B
Am I missing something?
Subject
vs
/Subject (Subject)
It seems like it should be fine.
-RickYou have a disconnect with regard to how Document Info fields are to be mapped to XMP Dublin Core (dc) data fields:
- Document Info Subject -> dc:description
[nobody claims this is intuitive, but it is indeed specified reasonably well]]
The following is not required in your case, but for your info I include it anyway:
- Document Info Keywords -> dc:subject
Your best source for this is the actual PDF/A-1 standard (ISO 19005-1, can be purchased form www.iso.org).
A very good companion for such type of topics are the TechNotes from the PDF/A Competence Center (see www.pdfa.org), downloadable free of charge.
HTH.
Olaf Drümmer -
CUCM 8.6 call busy between SIP phones and thirdparty phones
Hi Everybody,
I have the following error on my logs:
Invalid Disconnect Cause(cause=47), No Reason Header Appended
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getXCiscoViPRFallbackIDAndDTMFKey: Device type 8, Pstn Fallback is not enabled|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/compareAndUpdateMedia: sdpStatus=0, CMEndPointSDP role=1, SIPEndPointSdpRole=2|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPUACSessionExpires: isMidCall[0], response[200], method[102]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/parseSessionExpires: refresh_interval[1800], refresher[uas]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPSessionExpiresTimer: interval[1768] secs|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSecureRec: enforce srtp flag: 0, remote end srtp support: 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/updateCNToCC: identityCngFlag[0x1f], isConnInfoInd[1], ccContactHeader[]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPUpdateFlags: mIsUpdateForSignalingAllowed = 1 mIsUpdateForMediaAllowed = 1 mPendingOutgoingUpdate = 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/addTransparencyInfo: attaching transparency object|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefAe: SIPCdpc=281707, nodeId=3, processNumber=73 ci=144600614, branch=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |MRM::waiting_MrmDeallocateMtpResourceReq- Deallocate received for CI=53993831 count=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |MRM::waiting_DeallocateMtpResourceReq- ERROR Deallocate received for an unknown Call Identifier Ci = 53993831|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.0.15:[5060]:
the calling number is 34967850938
the callied number is 19026Julien,
Please use the link be low to collect cucm traces and use the advanced editor on the forum (located on top right hand corner of the discussion widnow) to attach the trace
https://supportforums.cisco.com/docs/DOC-29901
Ensure you collect the trace from the folowing
1. the server that the phone is registered to
2. If this server is different from the server in the cucm group of the sip trunk, then you need to also collect traces from the server (s) in the cucm group assiged to the sip trunk that connects to the 3rd party cluster...
NB: If you have three servers in the cucm group of the sip trunk, you have to collect the trace from all three servers. This is because calls are dsitributed in a round robin fashion to servers in a sip trunk...
FInally before you send the trace over, please ensure the calling and called numbers are present. Also include the time of the test call
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
SIP Inspection and dynamic port opening after re-invite
Platform: ASA 8.3(2)
Hello,
I have SIP devices along with SipTrunk and media endpoints. I am having issues with the ASA not dynamically opening (sip inspect enabled) UDP ports for RTP after a SIP re-invite causes the media endpoints to change within SDP.
The problem as below.
Initial SIP invite setups properly with ports dynamically opened between the media endpoints in the ASA
Re-inivite from the SIP device causes the media endpoints to change within the SDP
ASA blocks ports associated to the new media endpoints
I can resolve this by allowing the ports in the ACL, but suprised this is not working as re-invites to change media endpoints is to be expected in SIP conversation.
Regards,
AJBelow is the script you can use to reproduce this. Points worth mentioning.
Initial invite sets up the media between SIP Trunk and a media device ( 10.1.2.150) in the inside network, SIP signalling will be with 10.1.2.100. At this poit RTP flows freely between the SIP Trunk and the media device.
If the call is fax, a re-invite will occur and this will cause the IP address to change in the SDP. The new media endpoint becomes 10.1.2.151 (This device is SIP and Media (T38) capable).
For every SIP call we establish 10.1.2.150 will be used for media, we do not want to change this behaviour.
ASA 8.3 (2)
conf t
interface Ethernet0/0
nameif Inside_Voice
security-level 100
ip address 10.1.2.11 255.255.255.0 standby 10.1.2.12
exit
interface Ethernet0/1
nameif Outside_SIP_Trunk
security-level 0
ip address 10.1.60.254 255.255.255.0 standby 10.1.60.253
exit
object-group network SIP_trunks
network-object 1.2.3.0 255.255.255.0
exit
object-group service SIP_service
service-object tcp destination eq sip
service-object udp destination eq sip
exit
object-group network SIP_inside_servers
network-object host 10.1.2.100
exit
access-list Outside_SIP_in extended permit object-group SIP_service object-group SIP_trunks object-group SIP_inside_servers
access-group Outside_SIP_in in interface Outside_SIP_Trunk
route Outside_SIP_Trunk 0.0.0.0 0.0.0.0 10.1.60.1
class-map inspection_default
match default-inspection-traffic
exit
policy-map global_policy
class inspection_default
inspect dns preset_dns_map
inspect ftp
inspect h323 h225
inspect h323 ras
inspect ip-options
inspect netbios
inspect rsh
inspect rtsp
inspect skinny
inspect esmtp
inspect sqlnet
inspect sunrpc
inspect tftp
inspect sip
inspect xdmcp
inspect icmp
inspect icmp error
lass class-default
set connection decrement-ttl
exit
service-policy global_policy global
end -
Sip trunk between CUCM7.0 and third party VOIP provider
Hi all,
I'm looking for a solution/howto configuration for setting up a SIP trunk between CUCM7.0 and a SIP-VoIP provider.
Got SIP username, password and SIP-proxy IP from the provider.
I've done such a setup on CUCME a couple of times, but never on the CUCM.
Who can put me on right way?
Can it be done on the CUCM, or must an IOS-Device be used (got a PSTN-GW connected through H323 with CUCM)?
THanks for the hint,
Greets NorbertHere we go.....
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
! Calling nr. incoming
voice translation-rule 40
rule 1 /\(.*\)/ /0\1/
! Discard prefix (calling nr.)
voice translation-rule 190
rule 1 /^0\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\1/
! Mapping, internat to external nr.
voice translation-rule 191
rule 10 /^[1-9].*/ /xxxxEXTERNALxxxx/
! for call-forwarding
rule 15 /^0\(.*\)/ /\1/
! Mapping external to internal nr.
voice translation-rule 192
rule 2 /^xxxxxEXTERNALxxxx/ /4xx/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2001 voip
corlist outgoing dialCORnoFax
description *** SIP-TRUNK (OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
max-conn 2
destination-pattern 9.T
session protocol sipv2
session target ipv4:2xx.xxx.xxx.xxx
session transport udp
! customer external nr. range (one dot at the and -> 0-9)
incoming called-number xxxxxxxx.
dtmf-relay rtp-nte
codec g711alaw
no vad
gateway
timer receive-rtp 1200
sip-ua
keepalive target ipv4:2xx.xxx.xxx.xxx
authentication username xxEXTERNAL NR.xxxxx password 7 111111111111111111111
calling-info pstn-to-sip from number set xxEXTERNAL NR.xxxxx
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar ipv4:2xx.xxx.xxx.xxx expires 60
host-registrar
Greets,
Norbert
Hope this help......Please rate if helpful -
Send DTMF with SIP INFO (c2600) configuration question
I have a cisco 2600 with VIC-2FXS port as VOIP Gateway, connecting to SIP Server to receive SIP Incoming calls. I am able to receive call and the vocie has been pass through both way; and I would like the 2600 send DTMF as SIP info but was not able to do so. I have ios 12.3, and from this configuration guide http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1048824 , it does not require any config for SIP info. I must missing something here, please advice. Thanks.
The config is following -
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
hostname abc
boot-start-marker
boot-end-marker
enable secret 5 xx
enable password xx
memory-size iomem 10
no aaa new-model
ip subnet-zero
no ip routing
no ip cef
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
no ip route-cache
full-duplex
no ip http server
ip classless
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
destination-pattern 1.T
session protocol sipv2
session target ipv4:192.168.1.224:5061
session transport udp
codec g711ulaw
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
line con 0
line aux 0
line vty 0 4
login
endI tried to set it, and for IOS 12.3(26) - the latest for 2610 - which dose not have that option. I use dtmf-relay rtp-nte instead; but it did not send RFC2833 event. From ethereal, no OOB events. It seems that the config I have does not have OOB DTMF enable; I compare the config I have with other examples but can not found anything wrong. Any suggestion, and what debug message I should enable, that may help to identify the issue.
Thanks.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
full-duplex
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
description Outbound Calls
destination-pattern 1.T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
session transport udp
dtmf-relay rtp-nte
no vad
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
dial-peer voice 100 pots
destination-pattern 8...
port 1/1/0
forward-digits 3
dial-peer voice 20 voip
description Incoming calls from PBX
incoming called-number .T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
dtmf-relay rtp-nte
no vad -
Problems between an UC520 and Asterisk with sip trunk
I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Tabla normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David. -
I sync all songs, photos and videos between my computer and iPod classic with no problems, but when I get to sync info I get this message: "iTunes cannot sync Calendar and contacts to iPod. Try logging out of Windows adn then logging back in". Could you please help?
Correct. When you update via iTunes all synced media that is not in your iTunes library will be lost.
As IO said before:
You can redownload most iTunes pruchases by:
Downloading past purchases from the App Store, iBookstore, and iTunes Store
I do not think it included audio books. -
I want to share info between my iPad and iPhone only but still share apps with my spouse. How do I configure that?
Need some more info - what info do you want to share?
-
Difference between source list and info records
Dear All,
What is the difference between Source List and Info Records?
Thanks N Regards
AnandasivaHi,
The source list is the list which shows the sources from which you can procure the material. If you have ticked the source list requirement in the material master, then you can procure the material only from the vendors mentioned in the source list.
In Inf Record, you maintain the data for a material supplied by a vendor. e.g. Material price, tax code, other conditions,, buyer code, delivery & payment terms.
Hope your requirement is fulfilled.
Reward points if the answer is helpful.
Regards,
Prashant Kolhatkar -
how to send SIP info DTMF to third party call control.
wow - a wealth of info!
Please supply some deatils, topology, etc. -
Difference between 3.5 and 7.0 cubes
Please help me understand the difference between 3.5 and 5.0 cubes.
Thanks,
ManasaHi All,
I am sorry, what is difference between 3.5 and 7.0 cubes.
Thanks. -
My contact info keeps changing between my iPhone and iPad on its on.
For some reason my contact info changes on its own between my iPhone and iPad. I can add a new photo and contact info and then
For some reason I look the next day and the pictures gone and contact info isn't there?well the way icloud works besides the back up part of it is if you add a contact on your phone it will go to any other device that has the same apple id on it, Why the info is change im not sure about that but you can try turning off icloud and see if it still happens
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After my mail was moved to iCloud, my mail app emails received have 23 lines of "routing data between the address and the message how can Imget rid of this "data"?
Found the answer a Mail>preferances>Viewing>Show header detail. Changed it to Default.
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DTMF issues on SIP trunk to Verizon
Were you able to resolve this problem? I am having an identical issue also with Verizon.
Our topology and symptoms are as follows:
Outside phone -> PSTN -> Vzn SBC -> Vzn SIP trunk -> CUBE -> CUCM / VM system
DTMF tones generated by an IP phone are heard and recognized by an outside (off-net) phone/system as you would expect. However, DTMF tones generated by an outside (off-net) phone are not recognized by our voice mail system. When listening to the DTMF tone on an IP phone, it sounds very distorted and faint. A sniffer trace performed on the CUBE shows RFC 2833 NTEs being received from Verizon, and they appear to be properly relayed by the CUBE to the destination. Payload type negotiated for both legs is 101.
We are running CUCM 6.1.5. We have a CUBE router between CUCM and the Verizon SIP trunk. The CUBE router is running 12.4(24)T3 with the IPIPGW feature set. Our voice mail system is an AVST CallXpress system running v7.9 software. To CUCM the AVST voice mail ports appear as DNs assigned to several SCCP 7940 phones (DNs are part of a hunt group, hunt pilot = vm pilot). The AVST masquerades and registers as the 7940 phones.
I tried applying the "dtmf-interworking rtp-nte" both globally and at the dial-peer level with no success. Attached is the debug output you suggested.
Maybe you are looking for
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My computer died, so trying to connect my phone on a new computer and download all my music to the new computer/itunes?
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Will the informations be recorded in the alert.log file? -----No.168
will the informations about the loss of a temporary file be recorded in the alert.log file?
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Everything is working well, except a couple of small issues 1. cluvfy stage -post nodeadd reports warnings that CRS is not installed on some nodes. Not sure why this is, everything seems to be working. 2. crs_stat -t doesn't return node db instances.
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Received an invalid security certificate warning from a new site launched on Business Catalyst
Just launched a new site on Business Catalyst. When checking online for the first time I received the following warning in my browser " uses an invalid security certificate. The certificate is only valid for the following names: *.worldsecuresystems
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Powerline adapters not working
On returning from holiday Monday afternoon I found that, although I could flick through the TV channels, there was no program information being displayed. I powered down everything and restarted it. Eventually an error message was displayed (C01). I