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Today's Topics:
1. Re: R: DTMF outbound Issue un CUBE (Divin John (dijohn))
2. Re: Cheap Router for 911 (Nick Matthews)
3. R: R: DTMF outbound Issue un CUBE (Alessandro Bertacco)
4. Re: R: R: DTMF outbound Issue un CUBE (Divin John (dijohn))
5. Re: R: R: DTMF outbound Issue un CUBE (Somphol Boonjing)
6. R: R: R: DTMF outbound Issue un CUBE (Alessandro Bertacco)
7. R: R: R: DTMF outbound Issue un CUBE (Alessandro Bertacco)
8. Re: R: R: R: DTMF outbound Issue un CUBE (Divin John (dijohn))
9. Re: R: R: R: DTMF outbound Issue un CUBE (Kenneth Hayes)
----------------------------------------------------------------------
Message: 1
Date: Sat, 28 Sep 2013 17:43:34 +0000
From: "Divin John (dijohn)" <[email protected]>
To: Alessandro Bertacco <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] R: DTMF outbound Issue un CUBE
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
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------------------------------
Message: 2
Date: Sat, 28 Sep 2013 23:03:16 -0400
From: Nick Matthews <[email protected]>
To: "Norton, Mike" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Cheap Router for 911
Message-ID:
<cam-k-npgxrtffmsyz5wbsqnm+t9-umq9ntm7gfngwqjqf89...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
There are also options like the 881V that can handle PSTN (and SRST
connections) that may be a good fit. Otherwise like others mentioned there
are E-911 and assorted other options.
-nick
On Fri, Sep 27, 2013 at 12:45 PM, Norton, Mike <[email protected]>wrote:
> Whether or not you can do this may depend on the terms of service with
> the provider. I know with PS-ALI service from Telus, simply forwarding the
> DIDs to ?someone? is not sufficient. Their tariff document for PS-ALI
> service specifically states that the DID must ring to the station that
> placed the 911 call.****
>
> ** **
>
> -- ****
>
> Mike Norton****
>
> I.T. Specialist****
>
> Peace Wapiti School Division No. 76****
>
> ** **
>
> *From:* cisco-voip [mailto:[email protected]] *On Behalf
> Of *Lelio Fulgenzi
> *Sent:* September-27-13 8:20 AM
>
> *To:* David Zhars
> *Cc:* [email protected]
> *Subject:* Re: [cisco-voip] Cheap Router for 911****
>
> ** **
>
>
> ok, from what I understand, you don't have to _physically_ send the 911
> call over that line for things to work. you just need and address attached
> to the number you're marking the calls with. here's how we do it.****
>
> - signed a waiver with our PSTN provider so they lift the filter and
> allow us to mark outbound calls with ANY number****
> - signed up for a PS-ALI service with another provider so we can enter
> any address for any DID****
> - DIDs are configured to ring through to our campus security****
>
> you can probably do the same with a 1FL installed at each building, and
> then forward the calls to a DID that is answered by someone. ****
>
> ** **
>
> not perfect, but it might buy you some time.****
>
> ** **
>
> not the answer you're looking for, but a different perspective.****
>
> ** **
>
> Lelio****
>
> ** **
>
> ** **
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst, Data Centre and Communications Facilities
> Computing and Communications Services (CCS)
> University of Guelph
>
> 519?824?4120 Ext 56354
> [email protected]
> www.uoguelph.ca/ccs
> Room 037, Animal Science and Nutrition Building
> Guelph, Ontario, N1G 2W1****
>
> ** **
> ------------------------------
>
> *From: *"David Zhars" <[email protected]>
> *To: *[email protected]
> *Sent: *Friday, September 27, 2013 9:10:04 AM
> *Subject: *[cisco-voip] Cheap Router for 911****
>
> Have a site with a 8 port 3560 router. Wired via fiber to another
> building with UCM 8.0.****
>
> What's a cheap little router I can add to this site for 911? Not worried
> about SRST, but proper address reporting for 911 would be nice!
>
> My VAR is recommending the 2901 which looks like total overkill.****
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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------------------------------
Message: 3
Date: Sun, 29 Sep 2013 10:50:48 +0200
From: "Alessandro Bertacco" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] R: R: DTMF outbound Issue un CUBE
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 19:44
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
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------------------------------
Message: 4
Date: Sun, 29 Sep 2013 08:57:05 +0000
From: "Divin John (dijohn)" <[email protected]>
To: Alessandro Bertacco <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] R: R: DTMF outbound Issue un CUBE
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
CUCM is dropping rtp-nte in the 200 OK.
Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To: <sip:[email protected]>
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1380444129
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 26
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 323
v=0
o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
s=SIP Call
c=IN IP4 192.168.97.230
t=0 0
m=audio 17050 RTP/AVP 18 0 8 101
b=AS:26
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To:
<sip:[email protected]>;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac7
5dc-21992748
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: <sip:[email protected]>
Remote-Party-ID:
<sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 193
v=0
o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
s=SIP Call
c=IN IP4 192.168.97.50
b=TIAS:64000
b=AS:64
t=0 0
m=audio 11078 RTP/AVP 0 ===> 101 is missing
b=AS:26
a=rtpmap:0 PCMU/8000
a=ptime:20
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 11:50 AM
To: "[email protected]" <[email protected]>
Cc: Divin John <[email protected]>
Subject: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 19:44
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
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------------------------------
Message: 5
Date: Sun, 29 Sep 2013 19:20:38 +1000
From: Somphol Boonjing <[email protected]>
To: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] R: R: DTMF outbound Issue un CUBE
Message-ID:
<CABcKtNGUvp3k9kkyGDmwAZb6LKn44YtgWvqOG9H=hgcacng...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
dtmf-relay on dial-peer voice 5 should be changed to 'rtp-nte'.
dial-peer voice 5 voip
description Voice CCM Imbound
preference 5
answer-address 799
destination-pattern 5..
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:192.168.97.100
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric <=== rtp-nte here.
Regards,
On Sun, Sep 29, 2013 at 6:57 PM, Divin John (dijohn) <[email protected]>wrote:
> CUCM is dropping rtp-nte in the 200 OK.
>
>
> Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
> Sent:
> INVITE sip:[email protected]:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
> From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
> To: <sip:[email protected]>
> Date: Sun, 29 Sep 2013 08:42:09 GMT
> Call-ID: [email protected]
> Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
> Min-SE: 1800
> Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
> CSeq: 101 INVITE
> Timestamp: 1380444129
> Contact: <sip:[email protected]:5060>
> History-Info: <sip:[email protected]:5060>;index=1
> Expires: 180
> Allow-Events: telephone-event
> Max-Forwards: 26
> Session-Expires: 1800
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 323
>
> v=0
> o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
> s=SIP Call
> c=IN IP4 192.168.97.230
> t=0 0
> m=audio 17050 RTP/AVP 18 0 8 101
> b=AS:26
> a=rtpmap:18 G729/8000/1
> a=fmtp:18 annexb=no
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
>
>
>
> Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
> From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
> To: <sip:[email protected]
> >;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac75dc-21992748
> Date: Sun, 29 Sep 2013 08:42:09 GMT
> Call-ID: [email protected]
> CSeq: 101 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> Supported: replaces
> Supported: X-cisco-srtp-fallback
> Supported: Geolocation
> Session-Expires: 1800;refresher=uas
> Require: timer
> P-Asserted-Identity: <sip:[email protected]>
> Remote-Party-ID: <sip:[email protected]
> >;party=called;screen=yes;privacy=off
> Contact: <sip:[email protected]:5060>
> Content-Type: application/sdp
> Content-Length: 193
>
> v=0
> o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
> s=SIP Call
> c=IN IP4 192.168.97.50
> b=TIAS:64000
> b=AS:64
> t=0 0
> m=audio 11078 RTP/AVP 0 ===> 101 is missing
> b=AS:26
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
> Regards,
> Divin
> From: Alessandro Bertacco <[email protected]>
> Date: Sunday, 29 September 2013 11:50 AM
> To: "[email protected]" <[email protected]>
> Cc: Divin John <[email protected]>
> Subject: R: R: [cisco-voip] DTMF outbound Issue un CUBE
>
> Hi all,****
>
> here the debug of an incoming calls, but the same issue happen in the
> outbound call.****
>
> ** **
>
> Thank you regards.****
>
> ** **
>
> Alessandro****
>
> ** **
>
> *Da:* Divin John (dijohn) [mailto:[email protected] <[email protected]>]
> *Inviato:* sabato 28 settembre 2013 19:44
> *A:* Alessandro Bertacco; [email protected]
> *Oggetto:* Re: R: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ** **
>
> Hi Alessandro,****
>
> ** **
>
> Try calling ****
>
> http://www.testcall.com/222-1111.html****
>
> ** **
>
> *Or an iVR to check.*****
>
> ** **
>
> *Regards,*****
>
> *Divin*****
>
> ** **
>
> *From: *Alessandro Bertacco <[email protected]>
> *Date: *Saturday, 28 September 2013 6:04 PM
> *To: *Divin John <[email protected]>, "[email protected]" <
> [email protected]>
> *Subject: *R: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ** **
>
> Hi John,****
>
> thank you for the answer.****
>
> Yes, I mean RFC-2833/rtp-nte.****
>
> ****
>
> At the moment I don't have the debug, but on Monday I'll capture ccsip
> mess from the cube and I'll post it.****
>
> Thank you.****
>
> ****
>
> Regards****
>
> Alessandro****
>
> ****
>
> ****
>
> ****
>
> *Da:* Divin John (dijohn) [mailto:[email protected] <[email protected]>]
> *Inviato:* sabato 28 settembre 2013 16:42
> *A:* Alessandro Bertacco; [email protected]
> *Oggetto:* Re: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hey Alessandro,****
>
> ****
>
> What do mean by in band? ****
>
> ****
>
> Inband RFC-2833/rtp-nte****
>
> ****
>
> Or****
>
> ****
>
> Inband PCM Audio (Raw tones in the G711 stream)****
>
> ****
>
> Do you have a debug ccsip mess from the CUBE?****
>
> ****
>
> Regards,****
>
> Divin****
>
> ****
>
> *From: *Alessandro Bertacco <[email protected]>
> *Date: *Saturday, 28 September 2013 5:22 PM
> *To: *"[email protected]" <[email protected]>
> *Subject: *[cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hi all,****
>
> I've a problem with DTMF relay only on outbound direction in this
> topology scenario:****
>
> ****
>
> SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
> <---SCCP---> IP Phone ****
>
> ****
>
> I'm using codec g711 with the provider that tell me to use in-band DTMF.**
> **
>
> ****
>
> SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound
> and inbound calls works fine, but DTMF relay works only from SIP provider
> to Cisco IP Phone.****
>
> ****
>
> ****
>
> From inside, any DTMF tone are relayed to the SIP provider.****
>
> ****
>
> For example, call from SCCP Phone to my mobile phone, from mobile phone I
> can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
> mobil ephone are recived to the internal SCCP phone.****
>
> ****
>
> This happen always the same in that two scenario:****
>
> Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
> UCM 8.6.X ---SCCP---> IP Phone****
>
> IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
> SIP ISP ----> Mobile Phone****
>
> ****
>
> Can anyone help me?****
>
> ****
>
> Thank you very much****
>
> ****
>
> Regards****
>
> ****
>
> Alessandro Bertacco****
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
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------------------------------
Message: 6
Date: Sun, 29 Sep 2013 13:09:18 +0200
From: "Alessandro Bertacco" <[email protected]>
To: <[email protected]>, <[email protected]>
Subject: [cisco-voip] R: R: R: DTMF outbound Issue un CUBE
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi Somphol,
that dial-peer is not involved in the scenario.
The dial-peer is for H323 communication that works fine also for DTMF.
The two dial peer involved in the scenario are two SIP dial-peer 2000 and
2001.
Thanks!
Alessandro
Da: cisco-voip [mailto:[email protected]] Per conto di
Somphol Boonjing
Inviato: domenica 29 settembre 2013 11:21
A: [email protected]
Oggetto: Re: [cisco-voip] R: R: DTMF outbound Issue un CUBE
Hi,
dtmf-relay on dial-peer voice 5 should be changed to 'rtp-nte'.
dial-peer voice 5 voip
description Voice CCM Imbound
preference 5
answer-address 799
destination-pattern 5..
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:192.168.97.100
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric <=== rtp-nte here.
Regards,
On Sun, Sep 29, 2013 at 6:57 PM, Divin John (dijohn) <[email protected]>
wrote:
CUCM is dropping rtp-nte in the 200 OK.
Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]
<mailto:sip%[email protected]> >;tag=3D9E9B60-1C8F
To: <sip:[email protected] <mailto:sip%[email protected]>
>
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1380444129
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 26
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 323
v=0
o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
s=SIP Call
c=IN IP4 192.168.97.230
t=0 0
m=audio 17050 RTP/AVP 18 0 8 101 <tel:18%200%208%20101>
b=AS:26
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]
<mailto:sip%[email protected]> >;tag=3D9E9B60-1C8F
To: <sip:[email protected] <mailto:sip%[email protected]>
>;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac75dc-21992748
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: <sip:[email protected]
<mailto:sip%[email protected]> >
Remote-Party-ID: <sip:[email protected] <mailto:sip%[email protected]>
>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 193
v=0
o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
s=SIP Call
c=IN IP4 192.168.97.50
b=TIAS:64000
b=AS:64
t=0 0
m=audio 11078 RTP/AVP 0 ===> 101 is missing
b=AS:26
a=rtpmap:0 PCMU/8000
a=ptime:20
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 11:50 AM
To: "[email protected]" <[email protected]>
Cc: Divin John <[email protected]>
Subject: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 19:44
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
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------------------------------
Message: 7
Date: Sun, 29 Sep 2013 14:01:41 +0200
From: "Alessandro Bertacco" <[email protected]>
To: "'Divin John \(dijohn\)'" <[email protected]>,
<[email protected]>
Subject: [cisco-voip] R: R: R: DTMF outbound Issue un CUBE
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Thank you Devin,
So the issue is on the CUCM Trunk configuration side?
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: domenica 29 settembre 2013 10:57
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: R: [cisco-voip] DTMF outbound Issue un CUBE
CUCM is dropping rtp-nte in the 200 OK.
Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To: <sip:[email protected]>
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1380444129
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 26
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 323
v=0
o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
s=SIP Call
c=IN IP4 192.168.97.230
t=0 0
m=audio 17050 RTP/AVP 18 0 8 101
b=AS:26
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To:
<sip:[email protected]>;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac7
5dc-21992748
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: <sip:[email protected]>
Remote-Party-ID:
<sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 193
v=0
o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
s=SIP Call
c=IN IP4 192.168.97.50
b=TIAS:64000
b=AS:64
t=0 0
m=audio 11078 RTP/AVP 0 ===> 101 is missing
b=AS:26
a=rtpmap:0 PCMU/8000
a=ptime:20
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 11:50 AM
To: "[email protected]" <[email protected]>
Cc: Divin John <[email protected]>
Subject: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 19:44
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
-------------- next part --------------
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URL:
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------------------------------
Message: 8
Date: Sun, 29 Sep 2013 13:48:18 +0000
From: "Divin John (dijohn)" <[email protected]>
To: Alessandro Bertacco <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] R: R: R: DTMF outbound Issue un CUBE
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Trunk/phone or CUCM for reason is ignoring the incoming RFC2833 cap
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 3:01 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Thank you Devin,
So the issue is on the CUCM Trunk configuration side?
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: domenica 29 settembre 2013 10:57
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: R: [cisco-voip] DTMF outbound Issue un CUBE
CUCM is dropping rtp-nte in the 200 OK.
Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To: <sip:[email protected]>
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1380444129
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 26
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 323
v=0
o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
s=SIP Call
c=IN IP4 192.168.97.230
t=0 0
m=audio 17050 RTP/AVP 18 0 8 101
b=AS:26
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To:
<sip:[email protected]>;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac7
5dc-21992748
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: <sip:[email protected]>
Remote-Party-ID:
<sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 193
v=0
o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
s=SIP Call
c=IN IP4 192.168.97.50
b=TIAS:64000
b=AS:64
t=0 0
m=audio 11078 RTP/AVP 0 ===> 101 is missing
b=AS:26
a=rtpmap:0 PCMU/8000
a=ptime:20
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 11:50 AM
To: "[email protected]" <[email protected]>
Cc: Divin John <[email protected]>
Subject: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 19:44
A: Alessandro Bertacco; [email protected]
Oggetto: Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
Or an iVR to check.
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 6:04 PM
To: Divin John <[email protected]>, "[email protected]"
<[email protected]>
Subject: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
Da: Divin John (dijohn) [mailto:[email protected]]
Inviato: sabato 28 settembre 2013 16:42
A: Alessandro Bertacco; [email protected]
Oggetto: Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Saturday, 28 September 2013 5:22 PM
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this topology
scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
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------------------------------
Message: 9
Date: Sun, 29 Sep 2013 10:24:47 -0400
From: Kenneth Hayes <[email protected]>
To: "Divin John (dijohn)" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] R: R: R: DTMF outbound Issue un CUBE
Message-ID: <3715499636967899559@unknownmsgid>
Content-Type: text/plain; charset="iso-8859-1"
This is true as well but what is the DTMF settings on your outbound
dial-peers?
Sent from my iPhone
On Sep 29, 2013, at 9:49 AM, "Divin John (dijohn)" <[email protected]> wrote:
Trunk/phone or CUCM for reason is ignoring the incoming RFC2833 cap
Regards,
Divin
From: Alessandro Bertacco <[email protected]>
Date: Sunday, 29 September 2013 3:01 PM
To: Divin John <[email protected]>, "[email protected]" <
[email protected]>
Subject: R: R: R: [cisco-voip] DTMF outbound Issue un CUBE
Thank you Devin,
So the issue is on the CUCM Trunk configuration side?
*Da:* Divin John (dijohn) [mailto:[email protected] <[email protected]>]
*Inviato:* domenica 29 settembre 2013 10:57
*A:* Alessandro Bertacco; [email protected]
*Oggetto:* Re: R: R: [cisco-voip] DTMF outbound Issue un CUBE
CUCM is dropping rtp-nte in the 200 OK.
Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To: <sip:[email protected]>
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 3705908658-0672731619-2994839540-2686971021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1380444129
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 26
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 323
v=0
o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166
s=SIP Call
c=IN IP4 192.168.97.230
t=0 0
m=audio 17050 RTP/AVP 18 0 8 101
b=AS:26
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487
From: "0125011302" <sip:[email protected]>;tag=3D9E9B60-1C8F
To: <sip:[email protected]
>;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac75dc-21992748
Date: Sun, 29 Sep 2013 08:42:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]
>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 193
v=0
o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100
s=SIP Call
c=IN IP4 192.168.97.50
b=TIAS:64000
b=AS:64
t=0 0
m=audio 11078 RTP/AVP 0 ===> 101 is missing
b=AS:26
a=rtpmap:0 PCMU/8000
a=ptime:20
Regards,
Divin
*From: *Alessandro Bertacco <[email protected]>
*Date: *Sunday, 29 September 2013 11:50 AM
*To: *"[email protected]" <[email protected]>
*Cc: *Divin John <[email protected]>
*Subject: *R: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi all,
here the debug of an incoming calls, but the same issue happen in the
outbound call.
Thank you regards.
Alessandro
*Da:* Divin John (dijohn) [mailto:[email protected] <[email protected]>]
*Inviato:* sabato 28 settembre 2013 19:44
*A:* Alessandro Bertacco; [email protected]
*Oggetto:* Re: R: [cisco-voip] DTMF outbound Issue un CUBE
Hi Alessandro,
Try calling
http://www.testcall.com/222-1111.html
*Or an iVR to check.*
*Regards,*
*Divin*
*From: *Alessandro Bertacco <[email protected]>
*Date: *Saturday, 28 September 2013 6:04 PM
*To: *Divin John <[email protected]>, "[email protected]" <
[email protected]>
*Subject: *R: [cisco-voip] DTMF outbound Issue un CUBE
Hi John,
thank you for the answer.
Yes, I mean RFC-2833/rtp-nte.
At the moment I don't have the debug, but on Monday I'll capture ccsip mess
from the cube and I'll post it.
Thank you.
Regards
Alessandro
*Da:* Divin John (dijohn) [mailto:[email protected] <[email protected]>]
*Inviato:* sabato 28 settembre 2013 16:42
*A:* Alessandro Bertacco; [email protected]
*Oggetto:* Re: [cisco-voip] DTMF outbound Issue un CUBE
Hey Alessandro,
What do mean by in band?
Inband RFC-2833/rtp-nte
Or
Inband PCM Audio (Raw tones in the G711 stream)
Do you have a debug ccsip mess from the CUBE?
Regards,
Divin
*From: *Alessandro Bertacco <[email protected]>
*Date: *Saturday, 28 September 2013 5:22 PM
*To: *"[email protected]" <[email protected]>
*Subject: *[cisco-voip] DTMF outbound Issue un CUBE
Hi all,
I've a problem with DTMF relay only on outbound direction in this
topology scenario:
SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
<---SCCP---> IP Phone
I'm using codec g711 with the provider that tell me to use in-band DTMF.
SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound and
inbound calls works fine, but DTMF relay works only from SIP provider to
Cisco IP Phone.
>From inside, any DTMF tone are relayed to the SIP provider.
For example, call from SCCP Phone to my mobile phone, from mobile phone I
can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
mobil ephone are recived to the internal SCCP phone.
This happen always the same in that two scenario:
Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
UCM 8.6.X ---SCCP---> IP Phone
IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
SIP ISP ----> Mobile Phone
Can anyone help me?
Thank you very much
Regards
Alessandro Bertacco
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