Re: CME --> SIP Trunk DTMF Issues

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

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Received on Thu Oct 01 2009 - 14:29:49 ART