Re: CME --> SIP Trunk DTMF Issues

UPDATE!!! So I called my other friends Asterisk PBX IVR and it works just
FINE! The odd thing is -- Obviously to get to my buddies IVR I am going
through my provider as well. Hmmmmmmmmmmmmm

On Fri, Oct 2, 2009 at 1:54 AM, Brad Ellis <brad_at_ccbootcamp.com> wrote:

> Which dtmf setting are you using? Post your config. If you want to
> randomly try a couple things, put this under your dial-peer:
>
> dtmf-relay rtp-nte
>
> if that doesn't work, try this:
>
> dtmf-relay h245-alphanumeric
>
> let me know if one of those work.
>
>
> thanks,
> Brad Ellis
> CCIE#5796 (R&S / Security)
> CCSI# 30482
> CEO / President
> CCBOOTCAMP - Cisco Learning Solutions Partner (CLSP)
> Email: brad_at_ccbootcamp.com
> Toll Free: 877-654-2243
> International: +1-702-968-5100
> Skype: skype:ccbootcamp?call
> FAX: +1-702-446-8012
> YES! We take Cisco Learning Credits!
> Training And Remote Racks: http://www.ccbootcamp.com
>
>
> -----Original Message-----
> From: nobody_at_groupstudy.com [mailto:nobody_at_groupstudy.com] On Behalf Of
> Joe Astorino
> Sent: Thursday, October 01, 2009 7:53 PM
> To: Nick Matthews
> Cc: Ryan West; Victor Cappuccio; Cisco certification
> Subject: Re: CME --> SIP Trunk DTMF Issues
>
> Hey Ryan,
>
> Yeah, I can hear fine...I can place and receive calls just fine, it is
> just
> DTMF not working. I am not certain what happens on the provider end,
> but
> the good news is one of my best friends helps admin that box so I can
> likely
> do as much debugging as we need on their end. I will let you guys know
> if I
> come up with a solution! In the meantime it sucks, because most of the
> time
> I use my home phone it is to call places like comcast or other utility
> places and they all have IVR : )
>
> On Thu, Oct 1, 2009 at 2:29 PM, Nick Matthews <matthn_at_gmail.com> wrote:
>
> > It may be time to start blaming whomever administers the Asterisk PBX.
> > If your debug shows your sending digits, your options are minimal.
> > Out-of-band (tones) is not a good solution as it requires transcoders
> > or a PSTN interface.
> >
> >
> > -nick
> >
> > On Thu, Oct 1, 2009 at 11:37 AM, Ryan West <rwest_at_zyedge.com> wrote:
> > > Joe,
> > >
> > >
> > >
> > > You can hear fine, right? Are you doing inspection on the router?
> Do
> > you
> > > know if your provider hairpins all calls through a proxy / SBC, or
> are
> > they
> > > allow direct connections to each of the endpoints? What does your
> > inbound
> > > ACL look like? SIP providers, for example, do not abide by the
> 16384
> > > 32767 range that Cisco uses.
> > >
> > >
> > >
> > > -ryan
> > >
> > >
> > >
> > > From: Joe Astorino [mailto:jastorino_at_ipexpert.com]
> > > Sent: Thursday, October 01, 2009 9:57 AM
> > > To: Ryan West
> > > Cc: Nick Matthews; Victor Cappuccio; Cisco certification
> > > Subject: Re: CME --> SIP Trunk DTMF Issues
> > >
> > >
> > >
> > > Here you are: I dialed the a toll-free number here and received the
> > > following output. When I began pressing numbers on the phone, I
> received
> > > nothing at all and hung up the call.
> > >
> > >
> > >
> > > Bono#debug ccsip messages
> > > SIP Call messages tracing is enabled
> > >
> > > *Oct 1 2009 06:58:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Sent:
> > > INVITE sip:18002662274_at_sip.myisp.net:5060 SIP/2.0
> > > Via: SIP/2.0/UDP 68.43.145.171:5060;branch=z9hG4bKDD0889
> > > Remote-Party-ID:
> > > <sip:xxxxxxxxxx_at_bono.mydomain.ORG <sip%3Axxxxxxxxxx_at_bono.mydomain.ORG>
> <sip%3Axxxxxxxxxx_at_bono.mydomain.ORG <sip%253Axxxxxxxxxx_at_bono.mydomain.ORG>
> >
> > >;party=calling;screen=no;privacy=off
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net>
> <sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>>
> > > Date: Thu, 01 Oct 2009 10:58:31 GMT
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > Supported: 100rel,timer,resource-priority,replaces
> > > Min-SE: 1800
> > > Cisco-Guid: 753103472-2913997278-2296680089-24166541
> > > User-Agent: Cisco-SIPGateway/IOS-12.x
> > > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> > SUBSCRIBE,
> > > NOTIFY, INFO, REGISTER
> > > CSeq: 101 INVITE
> > > Max-Forwards: 70
> > > Timestamp: 1254394711
> > > Contact: <sip:xxxxxxxxxx_at_68.43.145.171:5060>
> > > Expires: 180
> > > Allow-Events: telephone-event
> > > Content-Type: application/sdp
> > > Content-Disposition: session;handling=required
> > > Content-Length: 250
> > >
> > > v=0
> > > o=CiscoSystemsSIP-GW-UserAgent 8015 6976 IN IP4 68.43.145.171
> > > s=SIP Call
> > > c=IN IP4 68.43.145.171
> > > t=0 0
> > > m=audio 18892 RTP/AVP 0 101
> > > c=IN IP4 68.43.145.171
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:101 telephone-event/8000
> > > a=fmtp:101 0-16
> > > a=ptime:20
> > >
> > > Bono#
> > > *Oct 1 2009 06:58:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Received:
> > > SIP/2.0 100 Trying
> > > Via: SIP/2.0/UDP
> > > 68.43.145.171:5060;branch=z9hG4bKDD0889;received=68.43.145.171
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net>
> <sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>>
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > CSeq: 101 INVITE
> > > Server: Asterisk PBX 1.6.1.0
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces, timer
> > > Contact: <sip:18002662274_at_208.83.69.38<sip%3A18002662274_at_208.83.69.38>
> <sip%3A18002662274_at_208.83.69.38 <sip%253A18002662274_at_208.83.69.38>>>
> > > Content-Length: 0
> > >
> > >
> > >
> > > *Oct 1 2009 06:58:31: %SEC-6-IPACCESSLOGP: list firewall-wan denied
> udp
> > > 82.228.112.103(12991) -> 68.43.145.171(46307), 1 packet
> > > *Oct 1 2009 06:58:32: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Received:
> > > SIP/2.0 100 Trying
> > > Via: SIP/2.0/UDP
> > > 68.43.145.171:5060;branch=z9hG4bKDD0889;received=68.43.145.171
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net>
> <sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>>
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > CSeq: 101 INVITE
> > > Server: Asterisk PBX 1.6.1.0
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces, timer
> > > Contact: <sip:18002662274_at_208.83.69.38<sip%3A18002662274_at_208.83.69.38>
> <sip%3A18002662274_at_208.83.69.38 <sip%253A18002662274_at_208.83.69.38>>>
> > > Content-Length: 0
> > >
> > >
> > >
> > > *Oct 1 2009 06:58:32: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Received:
> > > SIP/2.0 183 Session Progress
> > > Via: SIP/2.0/UDP
> > > 68.43.145.171:5060;branch=z9hG4bKDD0889;received=68.43.145.171
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net> <
> sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>
> > >;tag=as251952a0
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > CSeq: 101 INVITE
> > > Server: Asterisk PBX 1.6.1.0
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces, timer
> > > Contact: <sip:18002662274_at_208.83.69.38<sip%3A18002662274_at_208.83.69.38>
> <sip%3A18002662274_at_208.83.69.38 <sip%253A18002662274_at_208.83.69.38>>>
> > > Content-Type: application/sdp
> > > Content-Lengt
> > > Bono#h: 263
> > >
> > > v=0
> > > o=root 1770246803 1770246803 IN IP4 208.83.69.38
> > > s=Asterisk PBX 1.6.1.0
> > > c=IN IP4 208.83.69.38
> > > t=0 0
> > > m=audio 17546 RTP/AVP 0 101
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:101 telephone-event/8000
> > > a=fmtp:101 0-16
> > > a=silenceSupp:off - - - -
> > > a=ptime:20
> > > a=sendrecv
> > >
> > > Bono#
> > > *Oct 1 2009 06:58:34: %SEC-6-IPACCESSLOGP: list firewall-wan denied
> udp
> > > 71.38.199.130(59459) -> 68.43.145.171(46307), 1 packet
> > > Bono#
> > > *Oct 1 2009 06:58:37: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Received:
> > > SIP/2.0 200 OK
> > > Via: SIP/2.0/UDP
> > > 68.43.145.171:5060;branch=z9hG4bKDD0889;received=68.43.145.171
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net> <
> sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>
> > >;tag=as251952a0
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > CSeq: 101 INVITE
> > > Server: Asterisk PBX 1.6.1.0
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces, timer
> > > Contact: <sip:18002662274_at_208.83.69.38<sip%3A18002662274_at_208.83.69.38>
> <sip%3A18002662274_at_208.83.69.38 <sip%253A18002662274_at_208.83.69.38>>>
> > > Content-Type: application/sdp
> > > Content-Length: 263
> > >
> > > v=0
> > > o=root 1770246803 1770246804 IN IP4 208.83.69.38
> > > s=Asterisk PBX 1.6.1.0
> > > c=IN IP4 208.83.69.38
> > > t=0 0
> > > m=audio 17546 RTP/AVP 0 101
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:101 telephone-event/8000
> > > a=fmtp:101 0-16
> > > a=silenceSupp:off - - - -
> > > a=ptime:20
> > > a=sendrecv
> > >
> > > Bono#
> > > *Oct 1 2009 06:58:37: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Sent:
> > > ACK sip:18002662274_at_208.83.69.38:5060 SIP/2.0
> > > Via: SIP/2.0/UDP 68.43.145.171:5060;branch=z9hG4bKDD11DD6
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net> <
> sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>
> > >;tag=as251952a0
> > > Date: Thu, 01 Oct 2009 10:58:31 GMT
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > Max-Forwards: 70
> > > CSeq: 101 ACK
> > > Allow-Events: telephone-event
> > > Content-Length: 0
> > >
> > >
> > > *Oct 1 2009 06:59:03: %SEC-6-IPACCESSLOGP: list firewall-wan denied
> udp
> > > 92.243.181.198(17959) -> 68.43.145.171(46307), 1 packet
> > > *Oct 1 2009 06:59:05: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Sent:
> > > BYE sip:18002662274_at_208.83.69.38:5060 SIP/2.0
> > > Via: SIP/2.0/UDP 68.43.145.171:5060;branch=z9hG4bKDD22473
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net> <
> sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>
> > >;tag=as251952a0
> > > Date: Thu, 01 Oct 2009 10:58:31 GMT
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > User-Agent: Cisco-SIPGateway/IOS-12.x
> > > Max-Forwards: 70
> > > Timestamp: 1254394745
> > > CSeq: 102 BYE
> > > Reason: Q.850;cause=16
> > > Content-Length: 0
> > >
> > >
> > >
> > > Bono#
> > > *Oct 1 2009 06:59:05: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> > > Received:
> > > SIP/2.0 200 OK
> > > Via: SIP/2.0/UDP
> > > 68.43.145.171:5060;branch=z9hG4bKDD22473;received=68.43.145.171
> > > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net> <
> sip%3Axxxxxxxxxx_at_sip.myisp.net <sip%253Axxxxxxxxxx_at_sip.myisp.net>>
> > >;tag=76A8690-17DB
> > > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_at_sip.myisp.net> <
> sip%3A18002662274_at_sip.myisp.net <sip%253A18002662274_at_sip.myisp.net>>
> > >;tag=as251952a0
> > > Call-ID: 30BA2A16-ADB011DE-88E98E99-170C08D_at_bono.mydomain.ORG
> > > CSeq: 102 BYE
> > > Server: Asterisk PBX 1.6.1.0
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces, timer
> > > Content-Length: 0
> > >
> > >
> >
>
>
>
> --
> Regards,
>
> Joe Astorino - CCIE #24347 R&S
> Technical Instructor - IPexpert, Inc.
> Cell: +1.586.212.6107
> Fax: +1.810.454.0130
> Mailto: jastorino_at_ipexpert.com
>
>
> Blogs and organic groups at http://www.ccie.net
>
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>
>

-- 
Regards,
Joe Astorino - CCIE #24347 R&S
Technical Instructor - IPexpert, Inc.
Cell: +1.586.212.6107
Fax: +1.810.454.0130
Mailto:  jastorino_at_ipexpert.com
Blogs and organic groups at http://www.ccie.net