Nicholas Matthews

Member Since: Dec 12, 2008

English
Nicholas Matthews commented on IPCC Outbound in "Not Ready" in Contact Center 6 years ago

Just for reference, the internal request to have this added is CSCti08057.  You can ask your...

Nicholas Matthews commented on CME 8.0 (15.0(1)XA) vs SIP trunk DTMF issues in IP Telephony 6 years ago

I would start by getting a packet capture.  Then, you can look if you receive the packets. ...

Nicholas Matthews commented on CME 8.0 (15.0(1)XA) vs SIP trunk DTMF issues in IP Telephony 6 years ago

You may want to try this command:voice service voip dtmf-interworking rtp-nteIt changes the...

Nicholas Matthews commented on Which Trace to turn on for h323 in CUCM? in IP Telephony 6 years ago

The source IP for that will be in hex. Ex: '0A0A0A01' is 10.10.10.1 in the H225 setup message as...

Nicholas Matthews commented on CallManager 6.1 SIP trunk to sr140 -brooktrout in IP Telephony 6 years ago

I see, MGCP for PSTN, SIP to Rightfax.MGCP T38 is hard to configure recently, due to some bugs....

Nicholas Matthews commented on internet down affecting calls? in IP Telephony 6 years ago

You may also want to make sure that the dial peers used for these types of calls are also not going...

Nicholas Matthews commented on CallManager 6.1 SIP trunk to sr140 -brooktrout in IP Telephony 6 years ago

Entirely confused - you say you used SIP but this is all MGCP configuration?-nick

Nicholas Matthews commented on cant hear the other party ringing in Other Collaboration, Voice, and Video Subjects 6 years ago

Those look correct, yes. Assuming you're using an E1 PRI.-nick

Nicholas Matthews commented on CallManager 6.1 SIP trunk to sr140 -brooktrout in IP Telephony 6 years ago

This may be helpful for you.-nick

Nicholas Matthews commented on Cisco 3911 + Asterisk in IP Telephony 6 years ago

FYI - the 3911 wasn't designed to work with 3rd party call control. The 7960/40 was the only phone...

Nicholas Matthews commented on FAX on the Standard ATA 186 in IP Telephony 6 years ago

Add network-clock-select 1 e1 0/0/0 (fit to your needs) if you have slips on your T1/E1 in show...

Nicholas Matthews commented on Sustained DTMF Tone in IP Telephony 6 years ago

CUCM has this service parameter that I've confirmed works with gateways:H225 DTMF DurationDefault...

Nicholas Matthews commented on CME + VG-224 in IP Telephony 6 years ago

You can use VG224 with SCCP with CME. You register it just like you would any other SCCP...

Nicholas Matthews commented on VG224 Issue with UCM7 & MGCP in IP Telephony 6 years ago

Are you sure you have all the correct dial peers? If you have 24 phones on this system, not all of...

Nicholas Matthews commented on Fast busy when calling extensions in same CSS in IP Telephony 6 years ago

debug mgcp packetdebug voip ccapi inoutdebug vpm signalon the fxs gateway."I call and get busy tone...

Nicholas Matthews commented on Fast busy when calling extensions in same CSS in IP Telephony 6 years ago

This sounds similar: CSCsr87229-nick

Nicholas Matthews commented on Eco on E1 gateway in IP Telephony 6 years ago

Echo can be hard to self diagnose. Generally you have to listen to the audio streams.You could be...

Nicholas Matthews commented on Understanding codec negotiation in IP Telephony 6 years ago

Those dial peers will not use anything except g729r8. (You could have accomplished the same thing...

Nicholas Matthews commented on Fast busy when calling extensions in same CSS in IP Telephony 6 years ago

What version on the gateway? What protocol?

Nicholas Matthews commented on No ringback when calling over Vitelity SIP trunk in IP Telephony 6 years ago

You should make sure that the codec you have chosen is being negotiated ( and available on the MTP...

Nicholas Matthews commented on No ringback when calling over Vitelity SIP trunk in IP Telephony 6 years ago

This isn't very easy to explain -Your H323 side is not doing fast start and you're getting media in...

Nicholas Matthews commented on IP2IP Gateway-Asterisk Integration in IP Telephony 6 years ago

You would need to look at the h323 side as well:debug h225 asn1debug h245 asn1debug h225 q931-nick

Nicholas Matthews commented on SIP trunk issue on CUCM 7! in IP Telephony 6 years ago

Hi Johan,CUCM can send an invitation with authentication credentials but cannot send a REGISTER...

Nicholas Matthews commented on Unity QSIG with ISR in Unified Communications Applications 6 years ago

You may want to go to cisco.com/interoperabilityThere are a ton of QSIG implementations that there...

Nicholas Matthews commented on SIP Service Provider to PSTN = Can it be done? in Video Over IP 6 years ago

The easiest way to imagine this is replacing your PRI with a SIP trunk. Unless you get into really...

Nicholas Matthews commented on FXSs proble with MGCP in IP Telephony 6 years ago

Likely your incoming dial peer isn't being matched correctly. 'debug voip dialpeer' should show a...

Nicholas Matthews commented on Sustained DTMF Tone in IP Telephony 6 years ago

If your gateway is H.323 then it should take effect - I've confirmed that in the lab before for...

Nicholas Matthews commented on Sip trunk problem on cme 3825 but works on AsteriksWin32 server in Other Collaboration, Voice, and Video Subjects 6 years ago

Normally when a router initiates a SIP request the source port will be a random port above 1024....

Nicholas Matthews commented on No ringback when calling over Vitelity SIP trunk in IP Telephony 6 years ago

Try removing these commands from your dial peer:progress_ind setup enable 3progress_ind alert...

Nicholas Matthews commented on Sustained DTMF Tone in IP Telephony 6 years ago

For H323 you will want to change it on the device sending the digits. The h245 methods use include...

Nicholas Matthews commented on Upgrading pvdm on 2811 router with adipservice 12.4(22)t in IP Telephony 6 years ago

Hi Hassan,It is possible to upgrade the DSPware for PVDMs. However, this is a last-resort tactic...

Nicholas Matthews commented on CUBE SIP to h323 in IP Telephony 6 years ago

The call is disconnecting with cause=47, which is no resources. This is usually because you don't...

Nicholas Matthews commented on cptone RU parameters??? in IP Telephony 6 years ago

I think this is what you're looking for:test voice tone RU showHere's output:nicmatth-sip#test...

Nicholas Matthews commented on Sip trunk problem on cme 3825 but works on AsteriksWin32 server in Other Collaboration, Voice, and Video Subjects 6 years ago

This would be the problem:Contact: If like Paolo said and you are behind NAT, you need to have SIP...

Nicholas Matthews commented on RTP Traffic Capture in Video Over IP 6 years ago

Looks right.sounds like another issue.-nick

Nicholas Matthews commented on CUCM registration of a WS-X6608 Module in IP Telephony 6 years ago

For those of you that may have similar problems, there is another bug with the same symptoms for...

Nicholas Matthews commented on Transfer/ fowarding in IP Telephony 6 years ago

Are you using G.729br8 with MGCP?If so, you're probably hitting this bug: CSCsy10653The workarounds...

Nicholas Matthews commented on xmedius fax ata fax in IP Telephony 6 years ago

You can configure both for the MGCP gateway, but they will not be able to talk directly to each...

Nicholas Matthews commented on QoS Benefits? in Voice over IP 6 years ago

There's a high likelihood that you do not need QoS for day-to-day operations.Implementing QoS in...

Nicholas Matthews commented on DS1 reset causes remaining DS1s to unregister - 2851 in IP Telephony 6 years ago

Haven't seen anything like this on recent IOS. If you're on 12.3 code up to about 12.4(11)T this...

Nicholas Matthews commented on Need help with Dial=peer in CISCO 2811 for sip in IP Telephony 6 years ago

Need more details to troubleshoot.Good chance it's a simple dial peer translation / caller id...

Nicholas Matthews commented on Missing commands for Telephony Service in IP Telephony 6 years ago

Incorrect. You're not on CME 4.3, you're on an older CME version.The most recent version is 12.4(...

Nicholas Matthews commented on SIP Provider calls to CUE disconnect at 30 seconds into message recording in Unified Communications Applications 6 years ago

The only reason you would need the voice-class source command is if your NAT wasn't working...

Nicholas Matthews commented on packet capture of voice traffic in IP Telephony 6 years ago

The 7940/60s do this automatically as a feature. It's called 'SPAN To PC Port'. On newer phones...

Nicholas Matthews commented on Sip trunk with cisco 2811 dont work! in Other Collaboration, Voice, and Video Subjects 6 years ago

If you're not getting debugs it means the SIP messages aren't processed by the router.Three primary...

Nicholas Matthews commented on Phone SIP 3911 with other SIP VoIP in Other Collaboration, Voice, and Video Subjects 6 years ago

Your best bet for a Cisco phone that will interop with other devices are 7940/60s. Most other...

Nicholas Matthews commented on PRI/SIP Trunk for Outbound Dialing in IP Telephony 6 years ago

Hi Jian,MGCP is a different case from H323. With SIP and H323, once the call is sent to the...

Nicholas Matthews commented on How to configure VG224 Analog gateway using MGCP in IP Telephony 6 years ago

This depends entirely on which of the above options you want to use. There are about 4 different...

Nicholas Matthews commented on SIP trunk issue in IP Telephony 6 years ago

Try these:voice service voipallow sip to sipno supp sip referno supp sip movedThese are general...

Bio

Cisco TAC - Multiservice Voice
CCIE RS, CCVP, CCIP
Interests:
SIP, MGCP, H323, CUBE, FXO/FXS, PRI, CAS, CME, CUE, DSPs, SCCP, RTP, SRTP, and various other 3 and 4 letter acronyms.










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