07-08-2009 07:01 AM - edited 03-15-2019 06:52 PM
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
07-08-2009 11:46 AM
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
07-10-2009 05:20 AM
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
07-10-2009 07:55 AM
On the trixbox side we see this information for the call:
edcapptrixs51*CLI>
* SIP Call1*CLI>
Curr. trans. direction: Incoming
Call-ID: 5ed764450aa9992146059aeb102652e0@10.71.1.84
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: f8661d80-a5714e70-ded5-a54ce0a@10.206.84.10
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:00104383161@10.206.84.10:5060
Caller-ID: 00104383161
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:00104383161@10.206.84.10:5060
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
How can this type be forced on the dial-peer?
Danny
07-13-2009 09:37 AM
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
07-13-2009 11:33 PM
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
07-14-2009 04:44 AM
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
07-16-2009 01:35 AM
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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