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Today's Topics:
1. Rightfax with CUCM 8.5 SIP (abbas Wali)
2. Unity Connection 9x Compose New message Dial by Name default?
(Jason Aarons (AM))
3. Re: Unity Connection 9x Compose New message Dial by Name
default? (Anthony Holloway)
4. R: R: R: DTMF outbound Issue un CUBE (Alessandro Bertacco)
5. RESOLVED: Unity Connection 9x Compose New message Dial by
Name default? (Jason Aarons (AM))
6. Re: Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
(Jason Aarons (AM))
7. SSH to SIP 6945 and debug to see what button they are
pressing? (Jason Aarons (AM))
8. Re: SSH to SIP 6945 and debug to see what button they are
pressing? (Stephen Welsh)
9. Cisco AQM (Michele Russo (AM))
10. Re: SSH to SIP 6945 and debug to see what button they are
pressing? (Anthony Holloway)
11. Re: SSH to SIP 6945 and debug to see what button they are
pressing? (Stephen Welsh)
12. Re: SSH to SIP 6945 and debug to see what button they are
pressing? (Ryan Ratliff (rratliff))
13. Re: Router as H323 GW and also as SIP GW (vignesh sethuraman)
14. Re: Router as H323 GW and also as SIP GW (Anthony Holloway)
15. Re: Router as H323 GW and also as SIP GW (vignesh sethuraman)
16. Re: Router as H323 GW and also as SIP GW (Anthony Holloway)
----------------------------------------------------------------------
Message: 1
Date: Wed, 2 Oct 2013 17:41:58 +0100
From: abbas Wali <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Rightfax with CUCM 8.5 SIP
Message-ID:
<CAFdHCp4rEVY+NW1S-bSLE30k9CvhpAmF4NCmk5w1-rRfOJT=w...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi
does anyone knows or guide how to configure Rightfax settings with CUCM 8.5
via a SIP connection. we have MGCP gateways.
currently I can receive faxes but outbound is dead. Cant see any traffic on
the SIP trunk on the outbound direction (testing internal faxes only at
this time)
any outbound fax comes up with phone line problem
-- ABW5244588C5829
-------------------------------------------------------------------------------------------------
15:54 26/09/2013 Transmission Record
Sent to: ss
Phone: 63241
Billing information: '', ''
Remote ID:
Unique ID: "ABW5244588C5829"
Elapsed time: 0 minutes, 2 seconds.
Used channel 2 on server "NCCHQ-FAXSERVER".
No ANI data.
No AOC data.
Resulting status code (0/329; 0/0): Special Info Tone
No pages sent.
not 100% sure what my Brooktrout config tool SIP setup should be ??
Just appears that the call doesnt even make out of the rightfax box ,
though i have got a "+" Dialling rule configured with the number
adjustment.
any help will be appreciated.
many thanks
--
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Message: 2
Date: Wed, 2 Oct 2013 13:03:13 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] Unity Connection 9x Compose New message Dial by
Name default?
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d3bc1b...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Anyone know a way to turn default the "Send a Message Press 2" to use Dial by
Extension by default and have to press ## to switch to dial by name?
Correct Procedure to send a Message in Unity Connection Voicemail
1. Login to Voicemail.
2. To send a Message Press 2.
<record message>
Press ## <<<<<<<<<<<<<--- This is the secret to change between Dial by
Name versus Dial by Extension , Unity also asks the user if they want to switch
to "number entry"
Enter the 7-digit extension you want to send the message to.
<<<<<<<<<<<<<<<<<<<<<<<<This is not the 4-digit extension on phone
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Message: 3
Date: Wed, 2 Oct 2013 12:20:37 -0500
From: Anthony Holloway <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] Unity Connection 9x Compose New message Dial
by Name default?
Message-ID:
<cacrcjogrmt+dfkcetuuo9jtgfm_otj0vzfh0j30jh8-o19x...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
You can do it per mailbox like this:
Edit > Send Message Settings > Enter a Recipient By
And of course that can be set on a User Template for use during creation
time, or via Bulk Edit for already created users in bulk fashion.
*Forgot to reply all for the benefit of the list, sorry for the duplicate
message Jason.
On Wed, Oct 2, 2013 at 12:03 PM, Jason Aarons (AM) <
[email protected]> wrote:
> Anyone know a way to turn default the ?Send a Message Press 2? to use Dial
> by Extension by default and have to press ## to switch to dial by name? **
> **
>
> ** **
>
> Correct Procedure to send a Message in Unity Connection Voicemail****
>
> ****
>
> **1. **Login to Voicemail.****
>
> **2. **To send a Message Press 2.****
>
> ****
>
> <record message>****
>
> ****
>
> Press ## <<<<<<<<<<<<<--- This is the secret to change between Dial
> by Name versus Dial by Extension , Unity also asks the user if they want to
> switch to ?number entry?****
>
> ****
>
> Enter the 7-digit extension you want to send the message to.
> <<<<<<<<<<<<<<<<<<<<<<<<This
> is not the 4-digit extension on phone****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 4
Date: Wed, 2 Oct 2013 21:11:55 +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.
Thank you for the answer, but the dial-peer you suggest don't solve my
issue.
I need to investigate CUCM trace and if possible evaluate IOS upgrade on the
CUBE.
Here the SIP trunk configuration snapshot from the cucm side.
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
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Message: 5
Date: Wed, 2 Oct 2013 16:07:05 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Anthony Holloway <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: [cisco-voip] RESOLVED: Unity Connection 9x Compose New
message Dial by Name default?
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d3bc1b...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="utf-8"
Spot on! Thanks -jason
From: [email protected] [mailto:[email protected]] On Behalf Of Anthony
Holloway
Sent: Wednesday, October 02, 2013 1:21 PM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected])
Subject: Re: [cisco-voip] Unity Connection 9x Compose New message Dial by Name
default?
You can do it per mailbox like this:
Edit > Send Message Settings > Enter a Recipient By
And of course that can be set on a User Template for use during creation time,
or via Bulk Edit for already created users in bulk fashion.
*Forgot to reply all for the benefit of the list, sorry for the duplicate
message Jason.
On Wed, Oct 2, 2013 at 12:03 PM, Jason Aarons (AM)
<[email protected]<mailto:[email protected]>> wrote:
Anyone know a way to turn default the ?Send a Message Press 2? to use Dial by
Extension by default and have to press ## to switch to dial by name?
Correct Procedure to send a Message in Unity Connection Voicemail
1. Login to Voicemail.
2. To send a Message Press 2.
<record message>
Press ## <<<<<<<<<<<<<--- This is the secret to change between Dial by
Name versus Dial by Extension , Unity also asks the user if they want to switch
to ?number entry?
Enter the 7-digit extension you want to send the message to.
<<<<<<<<<<<<<<<<<<<<<<<<This is not the 4-digit extension on phone
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
itevomcid
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Message: 6
Date: Wed, 2 Oct 2013 18:09:27 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "Haas, Neal" <[email protected]>, "cisco-voip
([email protected])" <[email protected]>
Subject: Re: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965
(CCM 9.1)
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d3bc1b...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
So we ran Wireshark on both sides and audio showed it was existing the
Informacast NIC and the Phone NIC showed it was listening.
We disabled IGMP snooping on the switch the server is connected to, which
resolved the playout issues.
We listed only the two Subscribers.
From: Haas, Neal [mailto:[email protected]]
Sent: Thursday, September 26, 2013 7:03 PM
To: Jason Aarons (AM); cisco-voip ([email protected])
Subject: RE: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
Just the ones with phones
Neal Haas
From: Jason Aarons (AM) [mailto:[email protected]]
Sent: Thursday, September 26, 2013 3:51 PM
To: Haas, Neal; cisco-voip
([email protected]<mailto:[email protected]>)
Subject: RE: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
We have a 10 server cluster. The PUB, MOH and TFTP servers don't run CCM.
Should I include them in my list of IP addresses? Or just the Subscribers doing
call processing?
Recipient group members updated
Unable to build recipient groups: java.lang.Exception: The Communications
Managers given for cluster 'UDEL' are not consistent with those in the SNMP
cluster table. Please verify that the Communications Managers belong in the
cluster.
From: Haas, Neal [mailto:[email protected]]
Sent: Thursday, September 26, 2013 6:38 PM
To: Jason Aarons (AM); Jason Aarons (AM); cisco-voip
([email protected]<mailto:[email protected]>)
Subject: RE: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
Try this.
In informaCast (I have ver 8.1)
Admin--> Telephony --> CUCM clusters, click edit. Under Communications Manager
IP Address(es), make sure all of your IP's of your sub and pubs are there.
Verify that your SNMP values are all in your subs and pub.
Go to the Recipients tab, click UPDATE
Wait 10 to 20 minutes then try it again....
I also noticed a warning that not all my phones in call manager were
discovered. "There are more phones associated with your Communications Manager
server than your InformaCast license key supports. Broadcast messages will be
limited to 50 total phones. The number of phones in the list that will
participate in a broadcast depends on how many other phones have been broadcast
participants. For example, if 50 other phones have been broadcast participants,
then no phones in the list can participate. Otherwise, either all or some of
the phones can participate." Translation: your MAX page(s) will be 50 phones,
you can have as many phones in the cluster as you want
Neal Haas
IT Analyst, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890
FAX (559) 455-4747
County of Fresno
Information Technology Services Division
1020 S Tenth St
Fresno, CA 93702
From: cisco-voip [mailto:[email protected]] On Behalf Of Jason
Aarons (AM)
Sent: Thursday, September 26, 2013 3:12 PM
To: Jason Aarons (AM); cisco-voip
([email protected]<mailto:[email protected]>)
Subject: Re: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
So Cisco TAC is stumped.
If I used a filter of Device Pool on a Informacast Recipient Group it worked.
Next hurdle is calling the CTI Route Point I hear;
Welcome to the Singlewire Informacast System...ding..ding
Were sorry no devices could be activated. Your broadcast will not be completed.
From: cisco-voip [mailto:[email protected]] On Behalf Of Jason
Aarons (AM)
Sent: Thursday, September 26, 2013 12:20 PM
To: cisco-voip ([email protected]<mailto:[email protected]>)
Subject: [cisco-voip] Informacast Basic Paging 8.3 to SIP 7965 (CCM 9.1)
Informcast 8.3 (with CallManger 9.1) isn't finding one of my SIP 7965s?
Several SIP 6945s are showing up.
Any known issues with SIP? I setup the CTI ports as SCCP, maybe it's mandatory
to setup a SIP Trunk instead??
I also noticed a warning that not all my phones in callmanager were discovered.
"There are more phones associated with your Communications Manager server than
your InformaCast license
key<https://10.16.32.43:8444/InformaCast/admin?cmd=manage-key&seq=pr:1f>
supports. Broadcast messages will be limited to 50 total phones. The number of
phones in the list that will participate in a broadcast depends on how many
other phones have been broadcast participants. For example, if 50 other phones
have been broadcast participants, then no phones in the list can participate.
Otherwise, either all or some of the phones can participate."
I know I'm limited to a broadcast group of 50 phones, but I didn't think the
size of the cluster mattered?
itevomcid
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Message: 7
Date: Thu, 3 Oct 2013 10:31:03 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] SSH to SIP 6945 and debug to see what button
they are pressing?
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d3bc1b...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Is it possible to SSH to SIP 6945 and debug to see what buttons the end user is
pressing? Trying to track down DTMF issue on SIP 6945.
CallManager 9.1
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Message: 8
Date: Thu, 3 Oct 2013 14:38:44 +0000
From: Stephen Welsh <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] SSH to SIP 6945 and debug to see what button
they are pressing?
Message-ID: <[email protected]>
Content-Type: text/plain; charset="windows-1252"
Hi Jason,
You can get in-call DTMF info via CTI (SIP or SCCP), it's a bit tricky to setup
but I can host a WebEx session to take you through it if you like?
Kind Regards.
Stephen Welsh
[cid:[email protected]]
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On 3 Oct 2013, at 15:31, "Jason Aarons (AM)"
<[email protected]<mailto:[email protected]>>
wrote:
Is it possible to SSH to SIP 6945 and debug to see what buttons the end user is
pressing? Trying to track down DTMF issue on SIP 6945.
CallManager 9.1
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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------------------------------
Message: 9
Date: Thu, 3 Oct 2013 10:49:21 -0400
From: "Michele Russo (AM)" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Cisco AQM
Message-ID:
<cb65a4d1d2e17640ba229cb811020f370f68a66...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Hi All,
I am having a bit of a problem with Cisco AQM and was hoping someone else has
run into a similar problem. We have AQM set up as a network recording solution
for a customer, we are able to collect the recordings but none of the users who
are set up with 'Evaluator or Manager' Roles can access the /wfo Web GUI.
Every user I have tried get the message 'Credentials are not correct please try
again'. The users are all licensed and the message I see in the logs is always
the same 'User <UCCX.Demo> was NOT authenticated' followed by a '401
UNAUTHORIZED'.
The only account I can get to authenticate is the local administrator account.
I have worked with Calabrio and we validated the LDAP integration. At this
point I am at a loss and so is the Calabrio Engineer.
Has anyone run across this problem and can offer any troubleshooting
suggestions??
Thanks.
Michele Russo
Consultant
Dimension Data NA
11730 Plaza America Drive Suite 350
Reston, Va 20190
202-460-3965 (cell)
571-203-4007 (desk)
[email protected]<mailto:[email protected]>
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Message: 10
Date: Thu, 3 Oct 2013 10:02:09 -0500
From: Anthony Holloway <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] SSH to SIP 6945 and debug to see what button
they are pressing?
Message-ID:
<cacrcjoih6v-acaaallefom+2mv-h0q0qm6vlhz3mqvdpv7u...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
I'll be interested to see who responds and with what info. We recently
where involved with Cisco TAC to figure out why so many users are dialing
911, and we were simply told that the CUCM traces/packet captures for
SIP/SCCP messages were the only way. Which we did see, but we also wanted
more low level debugs such as the phone firmware detecting physical button
presses.
As I am typing this reply, I see that Stephen has responded with an option,
and it looks to be call control based like I mentioned above.
On Thu, Oct 3, 2013 at 9:31 AM, Jason Aarons (AM) <
[email protected]> wrote:
> Is it possible to SSH to SIP 6945 and debug to see what buttons the end
> user is pressing? Trying to track down DTMF issue on SIP 6945.****
>
> ** **
>
> CallManager 9.1****
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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------------------------------
Message: 11
Date: Thu, 3 Oct 2013 15:12:52 +0000
From: Stephen Welsh <[email protected]>
To: Anthony Holloway <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] SSH to SIP 6945 and debug to see what button
they are pressing?
Message-ID: <[email protected]>
Content-Type: text/plain; charset="windows-1252"
Hi Anthony,
In developing our CTI Implementation inside PhoneView we have been able to
receive JTAPI DTMF events for in-session calls that includes the buttons
pressed during the call.
As I said it's tricky to do, and some of the information on how to do it I'd
class as UnifiedFX confidential as we may expose that capability in PhoneView
at some point ;)
However happy to spend some time on a WebEx showing nice guys like yourself and
Jason how to capture the details ;)
Kind Regards.
Stephen Welsh
[cid:[email protected]]
================================================================
Unified FX?s FREE Educational seminars on The Essentials of Endpoint Security &
Compliance
? Session 1: The Impact of Security by Default (Recording: http://goo.gl/2yJaKm)
? Session 2: Understanding and Managing ITL & CTL Files (Recording:
http://goo.gl/w05Dqh)
? Session 3: Leading Practices for Endpoint Security & Compliance (Recording:
http://goo.gl/GuXy2P)
NEW Educational Webinar Announced
"Drive to 9: Essential Upgrade Planning Considerations" (10th Oct)
"Phone Background Management & Branding" (24th Oct)
Spaces are limited and we anticipate high demand, so please register early:
http://events.unifiedfx.com<http://unifiedfx.createsend1.com/t/r-i-bhljyut-l-o/>
================================================================
On 3 Oct 2013, at 16:02, Anthony Holloway
<[email protected]<mailto:[email protected]>>
wrote:
I'll be interested to see who responds and with what info. We recently where
involved with Cisco TAC to figure out why so many users are dialing 911, and we
were simply told that the CUCM traces/packet captures for SIP/SCCP messages
were the only way. Which we did see, but we also wanted more low level debugs
such as the phone firmware detecting physical button presses.
As I am typing this reply, I see that Stephen has responded with an option, and
it looks to be call control based like I mentioned above.
On Thu, Oct 3, 2013 at 9:31 AM, Jason Aarons (AM)
<[email protected]<mailto:[email protected]>> wrote:
Is it possible to SSH to SIP 6945 and debug to see what buttons the end user is
pressing? Trying to track down DTMF issue on SIP 6945.
CallManager 9.1
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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------------------------------
Message: 12
Date: Thu, 3 Oct 2013 15:19:57 +0000
From: "Ryan Ratliff (rratliff)" <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] SSH to SIP 6945 and debug to see what button
they are pressing?
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="windows-1252"
For DTMF there will be a few debugs you may want to use.
"spy TSTN 1" - This will get you the keypadbutton press, and that a digit is
being sent. It won't get you which digit.
CP-6961] 0003858063 - TSTN> 'tstone_proc_msg_accu_info()'.
GG_SCCP_ID_KEYPAD_BUTTON, call:ln(26765048:1)...
0003858063 - TSTN> 'tstone_get_call_by_call_line()'. sccpmsg_call(26765048),
sccpmsg_line(1), success
0003858063 - ================================
0003858063 - TSTN> ACTION: sending single digit
0003858063 -
If you add "debug keyevent 1" then this will get the literal button press
events, again no info that I can tell as to which button.
[CP-6961] cbManageUsrDefKey(3420) dynamic for key
card id=talking
type="ondigit"
pParam=post?SACCU(digit_down)=%digit?SDialNum=@%digit
----
0003955385 - TSTN> 'tstone_proc_msg_accu_info()'. GG_SCCP_ID_KEYPAD_BUTTON,
call:ln(26765048:1)...
0003955385 - TSTN> 'tstone_get_call_by_call_line()'. sccpmsg_call(26765048),
sccpmsg_line(1), success
0003955385 - ================================
0003955385 - TSTN> ACTION: sending single digit
0003955385 -
cbManageUsrDefKey(3420) dynamic for key
card id=talking
type="ondigit_up"
pParam=post?SACCU(digit_up)=%digit
----
Adding "spy accu 1" gets you more useful info, but only while in a call.
0004055602 - ACCU: accu_proc_msg_mmi_cmd(). new mmi_cmd. cmd_id=5.
mmi_cmd_list_length=1
0004055602 - ACCU: accu_ep_event. idx=1, event=mmi_cmd_dial_digit
0004055602 - ACCU: accu_sccp_ep_evthd(). event=17(mmi_cmd_dial_digit)
0004055602 - ACCU: accu_sccp_ep_proc_mmi_cmd_dial_digit(). type=DOWN. key=2.
duration=0. not_play_tone=0 <-------
0004055602 - ACCU: accu_term_event. [E1 L5]. event=mmi_cmd_dial_digit.
state=CONNECTED
0004055602 - ACCU. accu_cmm_sig_kpad_echo(). cmm_term=0x0. digit=2. dir=1
duration=-1
0004055602 - ACCU: accu_sccp_keypad_button(). line_inst=1. call_ref=0x19866f8.
key=2
0004055602 - ACCU: accu_mmi_cmd_process(). free mmi cmd. cmd_id=5.
mmi_cmd_list_length=0
The keyevent debug is for the button presses. TSTN is the layer between the UI
and call control. ACCU is the actual call control, and this is most likely
where you will be interested.
If you want protocol level info then spy on 'PSCCP' or 'SIPCC' though don't
expect particularly friendly output.
-Ryan
On Oct 3, 2013, at 10:38 AM, Stephen Welsh
<[email protected]<mailto:[email protected]>> wrote:
Hi Jason,
You can get in-call DTMF info via CTI (SIP or SCCP), it's a bit tricky to setup
but I can host a WebEx session to take you through it if you like?
Kind Regards.
Stephen Welsh
<image001.png>
================================================================
Unified FX?s FREE Educational seminars on The Essentials of Endpoint Security &
Compliance
? Session 1: The Impact of Security by Default (Recording: http://goo.gl/2yJaKm)
? Session 2: Understanding and Managing ITL & CTL Files (Recording:
http://goo.gl/w05Dqh)
? Session 3: Leading Practices for Endpoint Security & Compliance (Recording:
http://goo.gl/GuXy2P)
NEW Educational Webinar Announced
"Drive to 9: Essential Upgrade Planning Considerations" (10th Oct)
"Phone Background Management & Branding" (24th Oct)
Spaces are limited and we anticipate high demand, so please register early:
http://events.unifiedfx.com<http://unifiedfx.createsend1.com/t/r-i-bhljyut-l-o/>
================================================================
On 3 Oct 2013, at 15:31, "Jason Aarons (AM)"
<[email protected]<mailto:[email protected]>>
wrote:
Is it possible to SSH to SIP 6945 and debug to see what buttons the end user is
pressing? Trying to track down DTMF issue on SIP 6945.
CallManager 9.1
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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------------------------------
Message: 13
Date: Thu, 3 Oct 2013 23:27:36 +0800 (SGT)
From: vignesh sethuraman <[email protected]>
To: Anthony Holloway <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
Dear All,
If I am using?H323 to SIP, do I need a CUBE License for this to be uploaded to
7206VXR Router. Here is my scenario
CUCM -----H.323 Trunk-------7206VXR-------SIP------- CUBE
Thanks,
Viki
________________________________
From: Anthony Holloway <[email protected]>
To: vignesh sethuraman <[email protected]>
Cc: "[email protected]" <[email protected]>
Sent: Tuesday, 1 October 2013 7:52 PM
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
I should have been more clear that there are a number of other commands you
should configure on your gateway as it pertains to SIP best practices and
interoperability, however I was simply stating the bare minimum command to
switch the protocol from H323 to SIP.
Example of other commands you may want to look into (not a comprehensive list):
voice service voip
?sip
? bind control source-interface <your interface>
? bind media source-interface <your interface>
dial-peer voice XXX voip
?dtmf-relay sip-notify rtp-nte digit-drop
Also, if you need to tweak SIP timers or retry counters you will need to use
the: sip-au section and then use the: session target sip-server command.
Hope that helps.
On Tue, Oct 1, 2013 at 12:35 PM, Anthony Holloway
<[email protected]> wrote:
A voice gateway has H.323 enabled by default, and as such, your voip dialpeers
are H323 unless you tell it otherwise.
>
>How you switch from H323 to SIP on a per dialpeer basis, is with the dialpeer
>command:? session protocol sipv2
>
>
>By doing it per dialpeer you are able to run in a mixed SIP/H323 environment.
>
>
>If you're going from TDM to SIP on your 7206, then that is all you need.? If
>you are doing H323 to SIP or SIP to SIP then you'll need the appropriate allow
>connection commands under voice service voip.? But since you mentioned CUBE, I
>assume you're already familiar with these commands.
>
>
>
>
>On Tue, Oct 1, 2013 at 10:21 AM, vignesh sethuraman <[email protected]>
>wrote:
>
>Hello Experts,
>>
>>
>>I have a 7206 router acting as H323 GW registered to a CUCM. Along with that,
>>I need to create a SIP trunk on the same router to a CUBE.
>>
>>
>>Please let me know if it is possible or not. And also do let me know if
>>something needs to be taken care.
>>
>>
>>Thanks,
>>Viki
>>_______________________________________________
>>cisco-voip mailing list
>>[email protected]
>>https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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------------------------------
Message: 14
Date: Thu, 3 Oct 2013 10:39:42 -0500
From: Anthony Holloway <[email protected]>
To: vignesh sethuraman <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
Message-ID:
<CACRCJOjxh+7Ca+dc2PTDaFy9owxp8Bk=yuh7y1-angpxgrv...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
CUBE is an IP-to-IP gateway, and so H323-to-SIP qualifies as such, and
therefore requires CUBE. However, CUBE licensines is honor based, so
technically you could run CUBE without the licenses.
On Thu, Oct 3, 2013 at 10:27 AM, vignesh sethuraman <
[email protected]> wrote:
> Dear All,
>
> If I am using H323 to SIP, do I need a CUBE License for this to be
> uploaded to 7206VXR Router. Here is my scenario
>
>
> CUCM -----H.323 Trunk-------7206VXR-------SIP------- CUBE
>
> Thanks,
> Viki
>
> ------------------------------
> *From:* Anthony Holloway <[email protected]>
> *To:* vignesh sethuraman <[email protected]>
> *Cc:* "[email protected]" <[email protected]>
> *Sent:* Tuesday, 1 October 2013 7:52 PM
> *Subject:* Re: [cisco-voip] Router as H323 GW and also as SIP GW
>
> I should have been more clear that there are a number of other commands
> you should configure on your gateway as it pertains to SIP best practices
> and interoperability, however I was simply stating the bare minimum command
> to switch the protocol from H323 to SIP.
>
> Example of other commands you may want to look into (not a comprehensive
> list):
>
> voice service voip
> sip
> bind control source-interface <your interface>
> bind media source-interface <your interface>
>
> dial-peer voice XXX voip
> dtmf-relay sip-notify rtp-nte digit-drop
>
> Also, if you need to tweak SIP timers or retry counters you will need to
> use the: sip-au section and then use the: session target sip-server command.
>
> Hope that helps.
>
>
> On Tue, Oct 1, 2013 at 12:35 PM, Anthony Holloway <
> [email protected]> wrote:
>
> A voice gateway has H.323 enabled by default, and as such, your voip
> dialpeers are H323 unless you tell it otherwise.
>
> How you switch from H323 to SIP on a per dialpeer basis, is with the
> dialpeer command: session protocol sipv2
>
> By doing it per dialpeer you are able to run in a mixed SIP/H323
> environment.
>
> If you're going from TDM to SIP on your 7206, then that is all you need.
> If you are doing H323 to SIP or SIP to SIP then you'll need the appropriate
> allow connection commands under voice service voip. But since you
> mentioned CUBE, I assume you're already familiar with these commands.
>
>
> On Tue, Oct 1, 2013 at 10:21 AM, vignesh sethuraman <
> [email protected]> wrote:
>
> Hello Experts,
>
> I have a 7206 router acting as H323 GW registered to a CUCM. Along with
> that, I need to create a SIP trunk on the same router to a CUBE.
>
> Please let me know if it is possible or not. And also do let me know if
> something needs to be taken care.
>
> Thanks,
> Viki
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
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------------------------------
Message: 15
Date: Thu, 3 Oct 2013 23:50:09 +0800 (SGT)
From: vignesh sethuraman <[email protected]>
To: Anthony Holloway <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
Hello Anthony,
Thanks alot for your prompt response.
My Router is running,?Cisco IOS Software, 7200 Software
(C7200P-ADVENTERPRISEK9-M), Version 12.4(24)T1, RELEASE SOFTWARE (fc3).
Is it possible to enable CUBE (With Honor based License) on 7206VXR
running?Version 12.4(24)T1?
Thanks,
Vik
________________________________
From: Anthony Holloway <[email protected]>
To: vignesh sethuraman <[email protected]>
Cc: "[email protected]" <[email protected]>
Sent: Thursday, 3 October 2013 5:39 PM
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
CUBE is an IP-to-IP gateway, and so H323-to-SIP qualifies as such, and
therefore requires CUBE.? However, CUBE licensines is honor based, so
technically you could run CUBE without the licenses.
On Thu, Oct 3, 2013 at 10:27 AM, vignesh sethuraman <[email protected]>
wrote:
Dear All,
>
>
>If I am using?H323 to SIP, do I need a CUBE License for this to be uploaded to
>7206VXR Router. Here is my scenario
>
>
>
>
>CUCM -----H.323 Trunk-------7206VXR-------SIP------- CUBE
>
>
>Thanks,
>Viki
>
>
>
>________________________________
> From: Anthony Holloway <[email protected]>
>To: vignesh sethuraman <[email protected]>
>Cc: "[email protected]" <[email protected]>
>Sent: Tuesday, 1 October 2013 7:52 PM
>Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
>
>
>
>I should have been more clear that there are a number of other commands you
>should configure on your gateway as it pertains to SIP best practices and
>interoperability, however I was simply stating the bare minimum command to
>switch the protocol from H323 to SIP.
>
>Example of other commands you may want to look into (not a comprehensive list):
>
>
>voice service voip
>?sip
>? bind control source-interface <your interface>
>? bind media source-interface <your interface>
>
>
>
>dial-peer voice XXX voip
>?dtmf-relay sip-notify rtp-nte digit-drop
>
>
>Also, if you need to tweak SIP timers or retry counters you will need to use
>the: sip-au section and then use the: session target sip-server command.
>
>
>Hope that helps.
>
>
>
>
>On Tue, Oct 1, 2013 at 12:35 PM, Anthony Holloway
><[email protected]> wrote:
>
>A voice gateway has H.323 enabled by default, and as such, your voip dialpeers
>are H323 unless you tell it otherwise.
>>
>>How you switch from H323 to SIP on a per dialpeer basis, is with the dialpeer
>>command:? session protocol sipv2
>>
>>
>>By doing it per dialpeer you are able to run in a mixed SIP/H323 environment.
>>
>>
>>If you're going from TDM to SIP on your 7206, then that is all you need.? If
>>you are doing H323 to SIP or SIP to SIP then you'll need the appropriate
>>allow connection commands under voice service voip.? But since you mentioned
>>CUBE, I assume you're already familiar with these commands.
>>
>>
>>
>>
>>On Tue, Oct 1, 2013 at 10:21 AM, vignesh sethuraman
>><[email protected]> wrote:
>>
>>Hello Experts,
>>>
>>>
>>>I have a 7206 router acting as H323 GW registered to a CUCM. Along with
>>>that, I need to create a SIP trunk on the same router to a CUBE.
>>>
>>>
>>>Please let me know if it is possible or not. And also do let me know if
>>>something needs to be taken care.
>>>
>>>
>>>Thanks,
>>>Viki
>>>_______________________________________________
>>>cisco-voip mailing list
>>>[email protected]
>>>https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>
>
>
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------------------------------
Message: 16
Date: Thu, 3 Oct 2013 10:52:47 -0500
From: Anthony Holloway <[email protected]>
To: vignesh sethuraman <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Router as H323 GW and also as SIP GW
Message-ID:
<CACRCJOhCJ=6cfsbmwhusnzk8wefikzykqecju5mu1uovurh...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Here's a good starting document on CUBE. It even covers IOS versions and
licensing:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_ch1_ps5640_TSD_Products_Configuration_Guide_Chapter.html
On Thu, Oct 3, 2013 at 10:50 AM, vignesh sethuraman <
[email protected]> wrote:
> Hello Anthony,
>
> Thanks alot for your prompt response.
>
> My Router is running, Cisco IOS Software, 7200 Software
> (C7200P-ADVENTERPRISEK9-M), Version 12.4(24)T1, RELEASE SOFTWARE (fc3).
>
> Is it possible to enable CUBE (With Honor based License) on 7206VXR
> running Version 12.4(24)T1?
>
> Thanks,
> Vik
> ****
>
> ------------------------------
> *From:* Anthony Holloway <[email protected]>
> *To:* vignesh sethuraman <[email protected]>
> *Cc:* "[email protected]" <[email protected]>
> *Sent:* Thursday, 3 October 2013 5:39 PM
>
> *Subject:* Re: [cisco-voip] Router as H323 GW and also as SIP GW
>
> CUBE is an IP-to-IP gateway, and so H323-to-SIP qualifies as such, and
> therefore requires CUBE. However, CUBE licensines is honor based, so
> technically you could run CUBE without the licenses.
>
>
> On Thu, Oct 3, 2013 at 10:27 AM, vignesh sethuraman <
> [email protected]> wrote:
>
> Dear All,
>
> If I am using H323 to SIP, do I need a CUBE License for this to be
> uploaded to 7206VXR Router. Here is my scenario
>
>
> CUCM -----H.323 Trunk-------7206VXR-------SIP------- CUBE
>
> Thanks,
> Viki
>
> ------------------------------
> *From:* Anthony Holloway <[email protected]>
> *To:* vignesh sethuraman <[email protected]>
> *Cc:* "[email protected]" <[email protected]>
> *Sent:* Tuesday, 1 October 2013 7:52 PM
> *Subject:* Re: [cisco-voip] Router as H323 GW and also as SIP GW
>
> I should have been more clear that there are a number of other commands
> you should configure on your gateway as it pertains to SIP best practices
> and interoperability, however I was simply stating the bare minimum command
> to switch the protocol from H323 to SIP.
>
> Example of other commands you may want to look into (not a comprehensive
> list):
>
> voice service voip
> sip
> bind control source-interface <your interface>
> bind media source-interface <your interface>
>
> dial-peer voice XXX voip
> dtmf-relay sip-notify rtp-nte digit-drop
>
> Also, if you need to tweak SIP timers or retry counters you will need to
> use the: sip-au section and then use the: session target sip-server command.
>
> Hope that helps.
>
>
> On Tue, Oct 1, 2013 at 12:35 PM, Anthony Holloway <
> [email protected]> wrote:
>
> A voice gateway has H.323 enabled by default, and as such, your voip
> dialpeers are H323 unless you tell it otherwise.
>
> How you switch from H323 to SIP on a per dialpeer basis, is with the
> dialpeer command: session protocol sipv2
>
> By doing it per dialpeer you are able to run in a mixed SIP/H323
> environment.
>
> If you're going from TDM to SIP on your 7206, then that is all you need.
> If you are doing H323 to SIP or SIP to SIP then you'll need the appropriate
> allow connection commands under voice service voip. But since you
> mentioned CUBE, I assume you're already familiar with these commands.
>
>
> On Tue, Oct 1, 2013 at 10:21 AM, vignesh sethuraman <
> [email protected]> wrote:
>
> Hello Experts,
>
> I have a 7206 router acting as H323 GW registered to a CUCM. Along with
> that, I need to create a SIP trunk on the same router to a CUBE.
>
> Please let me know if it is possible or not. And also do let me know if
> something needs to be taken care.
>
> Thanks,
> Viki
>
> _______________________________________________
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> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
>
>
>
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