[Asterisk-Dev] DTMF with ATA-186 devices and Cisco upstreams
Dave Wolven
dwolven at 123.net
Mon Jun 16 20:35:26 MST 2003
I use:
ATA186 -> Asterisk -> AS5300
DTMF works well using dtmfmode=inband for the AS5300, while my ATA186 is
left at default.
I have tried the following:
ATA186->Asterisk->AS5300->PSTN->AS5300
This too works.
Asterisk CVS-05/13/03-01:27:19
Hope this helps a little....
Thanks
Dave
On Mon, 2003-06-16 at 22:41, John Todd wrote:
> This bug/problem/issue is really starting to drive me crazy.
>
> DTMF outbound from SIP-based ATA-186 devices simply does not work when sent to other SIP devices that are Cisco-based. I have not tried with non-Cisco devices, so I don't know if this is specific to Cisco or not, but I am fairly sure it is specific to ATA-186 devices. I have almost no luck with getting Cisco end-units (either Iconnecthere's boxes, or my Cisco 3640's) to recognize DTMF sequences if they are originated with an ATA-186. Now, I may be completely off base here, but I was under the impression that RFC2833 DTMF sequences were intercepted by Asterisk, and re-generated when transmitted to the other channel. If that's the case, then why are some of them not making it through, even though it seems Asterisk can "see" those keypresses on the local channel? I would expect Asterisk to clean up and re-transmit all of these RFC2833 messages, so they would be interpreted exactly the same by the far-end device. That appears not to be the case. My CVS version is about f!
> our hours old.
>
>
>
> When terminating to Zap channels, sometimes the ATA-186 works... sometimes it doesn't. It may even change within a call - it will work fine for the first few key presses, then fail for all others. All keystrokes are shown correctly in the DEBUG output as "Sending pending DTMF", etc. I also have flaky performance with the 7960 via Zap channels, so this is even more mud in the water.
>
>
> Anyway, here's some more debug that I can predictably reproduce:
>
> ATA-186 -> SIP -> Asterisk -> SIP -> Iconnecthere :
> No DTMF gets through*
>
> 7960 -> SIP -> Asterisk -> SIP -> Iconnecthere :
> DTMF get through fine every time
>
> *=Maybe one keystroke out of 10 makes it through and is recognized
>
> In the output below, I see that there is extra debug in with the ATA-186 about "Difference is X, ms is Y" but I have no idea what that means, but it's the only difference I can see between the two.
>
> Now, both devices use the same Asterisk server and the same Iconnecthere account, I'm just using different UA's here on my desk. When I hit keys on the keypad on both units, I see these lines for each keypress:
>
> Cisco 7960:
> ...
> DEBUG[21521]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[21521]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 49 (1)
> DEBUG[21521]: File channel.c, Line 1115 (ast_read): Auto-deactivating generator
> DEBUG[21521]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[21521]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 49 (1)
> DEBUG[21521]: File channel.c, Line 1115 (ast_read): Auto-deactivating generator
> DEBUG[21521]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[21521]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 49 (1)
> ...
>
> Cisco ATA-186:
> ...
> DEBUG[22545]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[22545]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 51 (3)
> DEBUG[22545]: File rtp.c, Line 791 (ast_rtp_raw_write): Difference is 4072, ms is 529
> DEBUG[22545]: File channel.c, Line 1115 (ast_read): Auto-deactivating generator
> DEBUG[22545]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[22545]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 53 (5)
> DEBUG[22545]: File rtp.c, Line 791 (ast_rtp_raw_write): Difference is 4096, ms is 532
> DEBUG[22545]: File channel.c, Line 1115 (ast_read): Auto-deactivating generator
> DEBUG[22545]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
> DEBUG[22545]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 56 (8)
> DEBUG[22545]: File rtp.c, Line 791 (ast_rtp_raw_write): Difference is 4544, ms is 588
> DEBUG[22545]: File channel.c, Line 1115 (ast_read): Auto-deactivating generator
> ...
>
> sip.conf:
>
> ; The 7960
> [2203]
> type=friend
> username=2203
> secret=bungholiopassword
> host=dynamic
> mailbox=2203
> context=intern
> canreinvite=no
> dtmfmode=rfc2833
> nat=1
>
>
> ; The ATA-186
> [2204]
> type=friend
> username=2204
> secret=somesecretpassword
> mailbox=2203
> host=dynamic
> context=intern
> canreinvite=no
> dtmfmode=rfc2833
> nat=1
>
>
> ATA-186 settings for v2.16:
>
> UIPassword: *******_____________
> ToConfig: 0___________________
> UseTftp: 1___________________
> TftpURL: 172.16.0.0__________
> CfgInterval: 1800________________
> EncryptKey: ____________________
> Dhcp: 1___________________
> StaticIP: 10.0.1.20________
> StaticRoute: 10.0.1.1________
> StaticNetMask: 255.255.255.0_____
> UID0: 2205________________
> PWD0: ____________________
> UID1: 2204________________
> PWD1: ____________________
> GkOrProxy: 10.0.1.55_______
> Gateway: 0___________________
> GateWay2: 0.0.0.0_____________
> UseLoginID: 0___________________
> LoginID0: 0___________________
> LoginID1: 0___________________
> AltGk: 0___________________
> AltGkTimeOut: 0___________________
> GkTimeToLive: 300_________________
> GkId: .___________________
> UseSIP: 1___________________
> SIPRegInterval: 240_________________
> MaxRedirect: 5___________________
> SIPRegOn: 1___________________
> NATIP: 0.0.0.0_____________
> SIPPort: 5060________________
> MediaPort: 16384_______________
> OutBoundProxy: 0___________________
> NatServer: 0___________________
> NatTimer: 0x00000000__________
> LBRCodec: 0___________________
> AudioMode: 0x00140014__________
> RxCodec: 2___________________
> TxCodec: 2___________________
> NumTxFrames: 2___________________
> CallFeatures: 0xffffffff__________
> PaidFeatures: 0xffffffff__________
> CallerIdMethod: 0x00019e60__________
> FeatureTimer: 0x00000000__________
> Polarity: 0x00000000__________
> ConnectMode: 0x00060400__________
> AutMethod: 0x00000000__________
> TimeZone: 17__________________
> NTPIP: 10.0.1.1________
> AltNTPIP: 0.0.0.0_____________
> DNS1IP: 0.0.0.0_____________
> DNS2IP: 0.0.0.0_____________
> TOS: 0x000000a0__________
> SigTimer: 0x01418564__________
> OpFlags: 0x00000002__________
> VLANSetting: 0x0000002b________
> NPrintf: 0.0.0.0.0___________
> TraceFlags: 0x0000000f__________
> RingOnOffTime: 2,4,25______________
> IPDialPlan: 1___________________
> DialPlan: *St4-|#St4-|911|1>#t
> DialTone: 2,31538,30831,3100,3
> BusyTone: 2,30467,28959,1191,1
> ReorderTone: 2,30467,28959,1191,1
> RingBackTone: 2,30831,30467,1943,2
> CallWaitTone: 1,30831,0,5493,0,0,2
> AlertTone: 1,30467,0,5970,0,0,4
> CallCmd: Af;AH;BS;NA;CS;NA;Df....
>
>
> JT
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
More information about the asterisk-dev
mailing list