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>
> >;party=calling;screen=no;privacy=off
> > From: <sip:xxxxxxxxxx_at_sip.myisp.net <sip%3Axxxxxxxxxx_at_sip.myisp.net>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>>
> > 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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>>
> > 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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>>
> > 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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>>
> > 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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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>
> >;tag=76A8690-17DB
> > To: <sip:18002662274_at_sip.myisp.net <sip%3A18002662274_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