Outbound DTMF Issue

We are experiencing a problem with DTMF tones on external attendants. We have a CCM 4.2 cluster connected to a CUCM 8.6.1 via a QSIG ICT.
CCM 4.2 <--> ICT <--> CUCM 8.6 <--> CUBE <--> SIP
DTMF works OK from phones in the CUCM 8.6 cluster. All inbound DTMF works all the way through to the CCM 4.2 cluster. However, DTMF from phones in the CCM 4.2 cluster does not work. It was working OK last week when some changes (adding CUC integration) were made. I have tried reversing most of the changes, but cant get it working again.
Any ideas or TS steps recommended? Thanks!

Two things to add on this thread:
The statement "these are outbound calls - so they don't hit a VoIP peer." is not true. There is always an inbound and outbound dial-peer matched on IOS. If you do not have an inbound dial-peer to match, IOS will use dial-peer 0 which is almost always a bad thing. Make sure you have a VoIP dial-peer that will match for inbound calls from UCM to the router.
CCX does not support in-band DMTF such as RFC2833 with SIP. If you do not have OOB DTMF such as KPML (RFC4730) or H.245 alphanumeric, then you need to invoke an MTP for conversion.

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I updated firmware and now I can dial tones even from wireless phones. Use auto mode
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Hi All,
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Time : Nov 12 20:06:56.417 UTC
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ok..We need to use a different approach to resolve this..We need to prefix calls coming from cucm so as to break up the overlapping issue..
do this..
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there is no extra ship-to party.
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Hi Experts,
I am having issue with outbound IDoc generated from Purchase Order.(Message Type ORDER and Basic Type ORDERS05)
Issue is when i create PO idoc Segment E1EDP01 and field ACTION populates as 001 this works fine.
Now when made any changes to the purchase order.* PO trigers one more EDI - Idoc. but the ACTION field is stil as 002.*
Could any tell me how to fix this.
Cheers...Gopinath.

Hi,
Thank you.
I have found the issue and fixed.
Below three setting was missing same has been done in Partner Profile now ok.
Message type u2013 ORDCHG, Processing Code u2013 ME11 and Change flag On.
Cheers..Gopinath
Edited by: Gopinath A.R on Oct 8, 2009 3:14 PM

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Hi i have recently upgraded one of my branch office router from C876 to C897-VA-K9, and after 1 week i deployed the new router i started to see 1 gbps + traffic in my dialer 0 interface outbound in my MRTG. and after i restarted the router the graph looks normal for 5 to 6 days and detects gig traffic and the graph keeps moving up from 1 gbps+ to 2gbps + and so on and keeps up until restarted.
1# sh int dialer 0
Dialer0 is up, line protocol is up (spoofing)
Hardware is Unknown
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really appreciate if you can share any re

Node Bandwidth is 8 Mbps.
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Hi,
I am facing issue with unit of measure.
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Sorry.. Posting it in ABAP Development forum.

PO outbound interface - Issue with Unit of measure

Hi,
I am facing issue with unit of measure.
1) In the PO outbound idoc in segment E1EDP01, the unit is always converted into ISO code and sent. Is there any way to avoid it.
2) Now while changing unit to ISO code, it checks if it is preset in T006 table. Now in some of the cases like Unit FT3, there are no entries in T006 table, hence IDOC is going into error. Let me know how can this be resolved. Do I need to update T006 table if yes then how can I do it.
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Edited by: shahuraj shirure on Mar 5, 2008 1:06 AM

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I have a T1 on a 2851 that for some long distance calls will pass 10 digits to the provider but for others will pass only the first 7 digits.... I am out of ideas....
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dtmf-relay h245-alphanumeric
codec g711ulaw
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Sep 28 21:11:17.145: //-1/809690D99602/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x465B10E4; count=1
*Sep 28 21:11:17.149: //-1/809690D99602/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x465B10E4; count=0
*Sep 28 21:11:17.149: //-1/809690D99602/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x465B10E4; count=1
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*Sep 28 21:11:17.149: //-1/809690D99602/RXRULE/regxrule_dp_translate: calling_number=6261 calling_octet=0x1
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*Sep 28 21:11:17.149: //-1/809690D99602/RXRULE/regxrule_vp_translate: calling_number=6261 calling_octet=0x1
called_number=915027444121 called_octet=0xA1
redirect_number= redirect_type=4294967295 redirect_plan=4294967295
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*Sep 28 21:11:31.761: //-1/006881E29802/RXRULE/regxrule_dp_translate: calling_number=6261 calling_octet=0x1
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called_number=916304555261 called_octet=0xA1
redirect_number= redirect_type=4294967295 redirect_plan=4294967295

use the 'forward-digits-all' on your outbound dialPeers and use a voice translation rule to strip the 9 before sending to the PSTN.
the output of your translation rule debug is not enough and apparently does not seem to be used for your outbound calls, as the message states there is no profile assigned to the voice port/trunk group.
if you still have problems, post your entire config, minus sensative ip addresses, if you can.
also, run a 'debug isdn q931' on both tests and post that as well.
also, if this is an MGCP endpoint, then double-check the ccmAdmin>gateway configuration for the 'significant digits' field and post that if you like.
(it looks as if you're using the same PRI for all calls but trying to send LD down one channel and the rest down another)

Outbound goods issue IDOC

Hi
Which message type should i use to sent goods issue idoc to an external system?

HI
GOOD
KOMIM - CHAR1 - Transfer pick quantities to delivery / post goods issue (as of release 4.0)
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go through this link
http://www.sapgenie.com/sapedi/message_types_masterdata.htm
thanks
mrutyun

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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.
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DTMF Issues

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Hello Manish
Thanks for your reply. So here is the thing. It was just as i feared, by leaving this "REQUIRE MTP" check on the SIP trunk, now all video calls are being setup as Audio only. So this is not a good solution.
It would be nice is the MTP is only invoked for the audio only callers, from my reading this is how its supposed to work.
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ZyXel DSL modem (10MB download and 768Kbps or so upload)
DSL modem is operating in bridge mode
Cisco ASA 5505 in routed mode with ten users behind the ASA. Nothing fancy about the ASA setup.
Each user relies on their own FirePass or Cisco VPN client (outbound, no configuration required in the ASA) continously from 8am to 5pm. Outlook and light application usage over the VPN only.
On Friday, 05/17/13, the outbound connections were working well. Latency was good throughout the day (less than 40ms to Google). On the outside interface, output bandwidth was less than 500Kbps (much less for large portions of the day!). Three users were using streaming Internet radio.
On Monday, 05/20/13, the outbound connections were working poorly. Latency was bad (170ms and higher to Google). On the outside interface, output bandwidth was remaining steady throughout the day between 800Kbps and 850Kbps. Occasionally, the outside output bandwidth would drop to 40Kbps, then 600Kbps and then back to 800Kbps or so. No user on Monday was uploading any large files, no cloud backup or anything of the sort. No users were listening to Internet radio.
On Monday afternoon, I shut down five client machines and the outside output bandwidth was still around 800Kbps.
On Monday evening, I stopped by the office after each user had left the building and checked the outside interface output bandwidth and it was between 0Kbps and 45Kbps (virtually no load). I verified that all machines were powered on, but the VPN clients were disconnected.
On Friday and Monday, for the outside interface, the input bandwidth was between 200Kbps and 500Kbps with occasional higher spikes when users downloaded something.
What could cause the difference between Friday and Monday? Is the DSL upload simply maxed out? If the DSL upload is maxed out, why did it work well on Friday when I had a greater demand on the connection?
Thank you

Hello Modulus,
Did you take captures on the outside interface or the ASA ( This to check what is leaving the ASA ) as you are using VPN clients traffic will go encrypted so you will not be able to determine what is the traffic used for but at least you might notice some extra-traffic (outside the VPN traffic) as this does not look right or normal,
Regards
Julio

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I am looking at sproxy screen of ECC 6.0 system
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Hi,
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