DTMF between IOS gateway, Callmanager and Trixbox

Unanswered Question
Jul 8th, 2009

Hi guys,

I am trying to setup a Trixbox conference. This is working OK but when call from PSTN then the DTMF digits are not recognized. If I call from a SCCP phone which is registered to our callmanager it is working without a problem.

This is my setup:

Phone on PSTN <POTS>3825 12.4(15)T9<dail-peerH323>Callmanager 6.1<SIP>Trixbox<SIP>Sip-phone

I am using G711ulaw as codec, I am not using the media termation point checkbox in CUCM. The trunk is not working anymore if I do that.

All DTMF settings are set to RFC2833 (on extensions, sip trunk on Trixbox and callmanager).The dial peer is configured:

dial-peer voice 1234 voip

preference 1

destination-pattern XXXXXXXXXX

progress_ind setup enable 3

voice-class codec 1

session target ipv4:(ip addr CUCM)

dtmf-relay rtp-nte

fax rate disable

no vad

Does anyone has an idea what the problem is?

Thanks in advance,

Jeroen

I have this problem too.
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Nicholas Matthews Wed, 07/08/2009 - 11:46

Hi Jeroen,

You would want to check the payload type value in the SDP of the 200 OK/INVITE from the Trixbox. IOS will use 101 by default, but you can change this with a dial peer command "rtp payloard rtp-nte ".

-nick

jmaat Fri, 07/10/2009 - 05:20

Hi Nick,

Thanks a lot for your answer.

The H323 protocol is used between the router and Callmanager. A SIP trunk is used between Trixbox and Callmanager. Isn't it something between CUCM and Trixbox then?

I am not sure what is used on the Trixbox. I need to make sniffer trace but I need to go onsite. I will let you know the outcome.

Thanks!

Jeroen

dvdhouwen Fri, 07/10/2009 - 07:55

On the trixbox side we see this information for the call:

edcapptrixs51*CLI>

* SIP Call1*CLI>

Curr. trans. direction: Incoming

Call-ID: [email protected]

Owner channel ID:

Our Codec Capability: 12

Non-Codec Capability (DTMF): 1

Their Codec Capability: 0

Joint Codec Capability: 0

Format: 0x0 (nothing)

MaxCallBR: 0 kbps

Theoretical Address: 10.71.1.84:5060

Received Address: 10.71.1.84:5060

SIP Transfer mode: open

NAT Support: RFC3581

Audio IP: 10.200.4.201 (local)

Our Tag: as3a26fdb6

Their Tag: as1280e73a

SIP User agent: Asterisk PBX

Need Destroy: 0

Last Message: Rx: OPTIONS

Promiscuous Redir: No

Route: N/A

DTMF Mode: rfc2833

SIP Options: (none)

edcapptrixs51*CLI>

* SIP Call

Curr. trans. direction: Incoming

Call-ID: [email protected]

Owner channel ID: SIP/10.206.84.10-08221d80

Our Codec Capability: 4

Non-Codec Capability (DTMF): 1

Their Codec Capability: 4

Joint Codec Capability: 4

Format: 0x4 (ulaw)

MaxCallBR: 384 kbps

Theoretical Address: 10.206.84.10:5060

Received Address: 10.206.84.10:5060

SIP Transfer mode: open

NAT Support: RFC3581

Audio IP: 10.200.4.201 (local)

Our Tag: as3c0afcbc

Their Tag: 054c0f9d-8b3b-4e2d-912b-36db99c8b4d7-63064715

SIP User agent: Cisco-CUCM6.1

Peername: to-CCM-SIP-EU

Original uri: sip:[email protected]:5060

Caller-ID: 00104383161

Need Destroy: 0

Last Message: Rx: ACK

Promiscuous Redir: No

Route: sip:[email protected]:5060

DTMF Mode: rfc2833

SIP Options: replaces replace timer

-----------

The dial-peer on the router is configured with these settings:

dial-peer voice 4029 voip

preference 1

destination-pattern xxxxx4029

progress_ind setup enable 3

voice-class h323 1

session target ipv4:10.x00.1x.251

dtmf-relay rtp-nte h245-signal

fax rate disable

no vad

-------------

In this Troxbox forum thread is described that the expected payload type is 101 or 121

http://www.trixbox.org/forums/trixbox-forums/sip-and-iax-trunks-and-providers/rtp-payload-96-need-modify-source-code-and-re

How can this type be forced on the dial-peer?

Danny

Nicholas Matthews Mon, 07/13/2009 - 09:37

This output doesn't clarify whether or not it is 101 or 121 (or something else).

We will do 101 by default.

You can try this dial peer command:

rtp payload nte 121

hth,

nick

jmaat Mon, 07/13/2009 - 23:33

HI Nick,

Thanks for your answer This is the output of this command, it is still not working.

rotnar01(config-dial-peer)#rtp payload-type nte 121

ERROR: value 121 in use!

rotnar01(config-dial-peer)#rtp payload-type nte 101

rotnar01(config-dial-peer)#

Here are the properties of the dial-peer:

rotnar01#show dial-peer voice XXXX

VoiceOverIpPeerXXXX

peer type = voice, system default peer = FALSE, information type = voice,

description = `',

tag = XXXX, destination-pattern = `XXXXXXXXX',

voice reg type = 0, corresponding tag = 0,

allow watch = FALSE

answer-address = `', preference=1,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent

CLID Override RDNIS = disabled,

source carrier-id = `', target carrier-id = `',

source trunk-group-label = `', target trunk-group-label = `',

numbering Type = `unknown'

group = XXXX, Admin state is up, Operation state is up,

incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = enabled,

modem transport = system,

URI classes:

Incoming (Called) =

Incoming (Calling) =

Destination =

huntstop = disabled,

in bound application associated: 'DEFAULT'

out bound application associated: ''

dnis-map =

permission :both

incoming COR list:maximum capability

outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing):

incoming call blocking:

translation-profile = `'

disconnect-cause = `no-service'

advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4

type = voip, session-target = `ipv4:XX.XXX.XX.XX',

technology prefix:

settle-call = disabled

ip media DSCP = ef, ip signaling DSCP = af31,

ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41

ip video rsvp-fail DSCP = af41,

UDP checksum = disabled,

session-protocol = cisco, session-transport = system,

req-qos = best-effort, acc-qos = best-effort,

req-qos video = best-effort, acc-qos video = best-effort,

req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,

req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,

dtmf-relay = rtp-nte digit-drop,

RTP dynamic payload type values: NTE = 101

Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,

A-law=8, GSMAMR-NB=117 iLBC=116

h263+=118, h264=119

G726r16 using static payload

G726r24 using static payload

RTP comfort noise payload type = 19

fax rate = disable, payload size = 20 bytes

fax protocol = system

fax-relay ecm enable

Fax Relay SG3-to-G3 Enabled (by system configuration)

fax NSF = 0xAD0051 (default)

voice-class codec = 1

codec = g729r8, payload size = 20 bytes,

video codec = None

voice class codec = 1

text relay = disabled

Media Setting = flow-through (global)

Expect factor = 10, Icpif = 20,

Playout Mode is set to adaptive,

Initial 60 ms, Max 250 ms

Playout-delay Minimum mode is set to default, value 40 ms

Fax nominal 300 ms

Max Redirects = 1, signaling-type = cas,

VAD = disabled, Poor QOV Trap = disabled,

Source Interface = NONE

voice class sip url = system,

voice class sip rel1xx = system,

tvoice class sip outbound-proxy = system,

voice class sip asserted-id = system,

Thanks again,

Jeroen

Nicholas Matthews Tue, 07/14/2009 - 04:44

I see that you have rtp-nte digit drop configure. Please remove the digit drop as this is not a scenario where it is correct.

-nick

jmaat Thu, 07/16/2009 - 01:35

Hi Nick,

We already tried that. We tried all options. It's back now to dtmf-relay rtp-nte but still not working.

Thanks,

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