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AT&T vs Verizon DTMF recognition using OBi110 as an FXO Port on Asterisk

Started by otter117, July 19, 2011, 12:17:39 PM

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otter117

Using the directions provided in this forum I was able to setup an OBi110 as an FXO Port on Asterisk which is the platform we are providing an IVR application on.
My test unit was connected to an ILEC POTS line and all worked well.
My production units are attached to an ESI PBX where QWEST is the ILEC.
If I dial in with an AT&T iPhone or an AT&T Blackberry, DTMF recognition is good.
DTMF recognition using a Verizon Droid, LG or Samsung cell phone is awful.
I cannot interrupt the announcements from Asterisk and most of the time the OBi does not log a digit detected.
I have tried many of the DTMF tips I've seen in this forum such as adjusting gain levels, DTMFMethod, etc...
AT&T is TDMA or GSM, Verizon is CDMA. Is there a need to tweak the codec settings in the OBi110?
Is there a way to do a pass through of the DTMF from the Line Interface to Asterisk without any intervention by the OBi?
OBi110 FW: 1.2.1 (Build: 2384)

OBiSupport

Can you first test the device (OBi110) with a regular telephone and see if you can log all the digits? you may listen to it or open call status page, where "DTMF Digits Received" shows how many digits you received.

If you are using SP1 or SP2 to receive incoming calls, you can choose DTMF method to be "inband", so basically send DTMF tone through audio path to Asterisk directly, rather than OBi locally generates all digits, see if that works for you.

go to: Service Provider -> ITSP profile A/B -> DTMF method -> ...

Restore all gain settings back to default at this point, and see whether problem is caused by OBi or actually the gateway side.

otter117

Thanks for you reply.
Setting DTMFmethod to InBand did not help. It actually resulted in Asterisk not "hearing" digits from the iPhone even though they were logged by the Obi110 to my syslog. InBand also did not improve OBi110 recognition of Droid DTMF.
All gain values are set to the defaults.
I will try a "regular" phone when I get home.
Here are the entries in my syslog for the iPhone which shows a good call with all of the digits logged by the OBi110, heard and processed by Asterisk. DTMFmethod is auto.

I press 1 for the application.
1 for the language.
1 for the query type.
987654321# for the account
1 for the account type.
4321# for the password
Then I hang up. It works like I want it to.


Jul 19 14:07:18 10.70.5.205 Goodbye! Reboot Now. (reason: 4)
Jul 19 14:09:19 10.70.5.205 [PR] prompt transcoding done!!
Jul 19 14:13:19 10.70.5.205 [DAA]: FXO ring on
Jul 19 14:13:21 10.70.5.205
  • DAA CND 07191406,5555551212,Cell Phone  ,,,
    Jul 19 14:13:23 10.70.5.205 sendto a0a05cd:5060(996)
    Jul 19 14:13:23 10.70.5.205 INVITE sip:36226@10.10.5.205:5060 SIP/2.0  Call-ID: fefe4615@10.70.5.205  Content-Length: 400  CSeq: 8001 INVITE  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  Max-Forwards: 70  To: <sip:36226@10.10.5.205>  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-3d53be55;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384  Contact: <sip:obi110@10.70.5.205:5060>  Expires: 60  Supported: replaces  Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER  Remote-Party-ID: "Cell Phone  " <sip:5555551212@10.10.5.205>;party=calling;privacy=off  Content-Type: application/sdp    v=0  o=- 36160 1 IN IP4 10.70.5.205  s=-  c=IN IP4 10.70.5.205  t=0 0  m=audio 10033 RTP/AVP 0 8 18 104 102 103 105 101  a=rtpmap:0 PCMU/8000  a=rtpmap:8 PCMA/8000  a=rtpmap:18 G729/8000  a=rtpmap:104 G726-32/8000  a=rtpmap:102 G726-16/8000  a=rtpmap:103 G726-24/8000  a=rtpmap:105 G726-40/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-15  a=sendrecv  a=ptime:20  a=xg726bitorder:big-endian
    Jul 19 14:13:23 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 14:13:23 10.70.5.205 SIP/2.0 100 Trying  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-3d53be55;received=10.70.5.205;rport=5060  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  To: <sip:36226@10.10.5.205>  Call-ID: fefe4615@10.70.5.205  CSeq: 8001 INVITE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Contact: <sip:36226@10.10.5.205>  Content-Length: 0
    Jul 19 14:13:23 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 14:13:23 10.70.5.205 SIP/2.0 200 OK  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-3d53be55;received=10.70.5.205;rport=5060  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  To: <sip:36226@10.10.5.205>;tag=as2a5835f9  Call-ID: fefe4615@10.70.5.205  CSeq: 8001 INVITE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Contact: <sip:36226@10.10.5.205>  Content-Type: application/sdp  Content-Length: 211    v=0  o=root 15522 15522 IN IP4 10.10.5.205  s=session  c=IN IP4 10.10.5.205  t=0 0  m=audio 10026 RTP/AVP 0 101  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=ptime:20  a=sendrecv
    Jul 19 14:13:23 10.70.5.205 RTP:DtmfTxMtd:1(1),0
    Jul 19 14:13:23 10.70.5.205 sendto a0a05cd:5060(346)
    Jul 19 14:13:23 10.70.5.205 ACK sip:36226@10.10.5.205 SIP/2.0  Call-ID: fefe4615@10.70.5.205  Content-Length: 0  CSeq: 8001 ACK  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  Max-Forwards: 70  To: <sip:36226@10.10.5.205>;tag=as2a5835f9  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-77dde6df;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384
    Jul 19 14:13:23 10.70.5.205 [DAA]: FXO OFFHOOK
    Jul 19 14:13:23 10.70.5.205 [DAA]: detecting disconnect tone, reinit: 1
    Jul 19 14:13:23 10.70.5.205 RTP:Start->a0a05cd:10026(80);0;0;0:0:0;0(28)
    Jul 19 14:13:23 10.70.5.205 [DAA] FXO ring off
    Jul 19 14:13:27 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 359421 ----
    Jul 19 14:13:27 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:27 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 359681 ms----
    Jul 19 14:13:29 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 361881 ----
    Jul 19 14:13:29 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:30 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 362131 ms----
    Jul 19 14:13:33 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 365921 ----
    Jul 19 14:13:34 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:34 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 366171 ms----
    Jul 19 14:13:38 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 9, @ 370871 ----
    Jul 19 14:13:38 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 9, c_dgt:
    Jul 19 14:13:39 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 371431 ms----
    Jul 19 14:13:39 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 8, @ 371811 ----
    Jul 19 14:13:39 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 8, c_dgt:
    Jul 19 14:13:40 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 372151 ms----
    Jul 19 14:13:40 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 7, @ 372531 ----
    Jul 19 14:13:40 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 7, c_dgt:
    Jul 19 14:13:40 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 372891 ms----
    Jul 19 14:13:41 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 6, @ 373231 ----
    Jul 19 14:13:41 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 6, c_dgt:
    Jul 19 14:13:41 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 373571 ms----
    Jul 19 14:13:42 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 5, @ 373971 ----
    Jul 19 14:13:42 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 5, c_dgt:
    Jul 19 14:13:42 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 374311 ms----
    Jul 19 14:13:42 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 4, @ 374691 ----
    Jul 19 14:13:42 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 4, c_dgt:
    Jul 19 14:13:43 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 375011 ms----
    Jul 19 14:13:43 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 3, @ 375391 ----
    Jul 19 14:13:43 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 3, c_dgt:
    Jul 19 14:13:43 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 375731 ms----
    Jul 19 14:13:44 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 2, @ 376111 ----
    Jul 19 14:13:44 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 2, c_dgt:
    Jul 19 14:13:44 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 376471 ms----
    Jul 19 14:13:44 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 376831 ----
    Jul 19 14:13:44 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:45 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 377171 ms----
    Jul 19 14:13:45 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : #, @ 377571 ----
    Jul 19 14:13:45 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: #, c_dgt:
    Jul 19 14:13:45 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 377891 ms----
    Jul 19 14:13:50 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 382111 ----
    Jul 19 14:13:50 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:50 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 382391 ms----
    Jul 19 14:13:56 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 4, @ 388901 ----
    Jul 19 14:13:56 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 4, c_dgt:
    Jul 19 14:13:57 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 389191 ms----
    Jul 19 14:13:57 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 3, @ 389551 ----
    Jul 19 14:13:57 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 3, c_dgt:
    Jul 19 14:13:57 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 389911 ms----
    Jul 19 14:13:58 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 2, @ 390271 ----
    Jul 19 14:13:58 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 2, c_dgt:
    Jul 19 14:13:58 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 390631 ms----
    Jul 19 14:13:59 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 391011 ----
    Jul 19 14:13:59 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 14:13:59 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 391331 ms----
    Jul 19 14:13:59 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : #, @ 391731 ----
    Jul 19 14:13:59 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: #, c_dgt:
    Jul 19 14:14:00 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 392091 ms----
    Jul 19 14:14:12 10.70.5.205 t_on: 250, t_off: 250, steady_on: 280, steady_off: 320, hits: 5
    Jul 19 14:14:12 10.70.5.205 [DSP]: fxo tone detected: reorder, 3!
    Jul 19 14:14:12 10.70.5.205 [DSP]: fxo (disconnect) tone detected: reorder, 3!
    Jul 19 14:14:12 10.70.5.205
  • DAA Disc.Tone 0 0
    Jul 19 14:14:12 10.70.5.205 sendto a0a05cd:5060(429)
    Jul 19 14:14:12 10.70.5.205 BYE sip:36226@10.10.5.205 SIP/2.0  Call-ID: fefe4615@10.70.5.205  Content-Length: 0  CSeq: 8002 BYE  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  Max-Forwards: 70  To: <sip:36226@10.10.5.205>;tag=as2a5835f9  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-52262f4b;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384  X-RTP-Stat: PS=2487,OS=363180,PR=1578,OR=271416,PL=0,JI=1,DU=49,EN=G711U,DE=G711U
    Jul 19 14:14:12 10.70.5.205 RTP:Del Channel
    Jul 19 14:14:12 10.70.5.205 [JB] call overall status --       peer:             10.10.5.205:10026, local:            10.70.5.205:10033, pkt_tx:           2487, pkt_rx:           1578, bytes_tx:         363180, bytes_rx:         271416, clk_diff:         0 PPM, pkt_in_jb:        0, pkt_ooo:          0, pkt_lost:         0, pkt_late:         0, pkt_loss_rate:    0 %, pkt_drop_rate:    0 %, jb_len:           190 ms, curr_rcvd_jitter: 1 ms, rcvd_digits:      0, underruns:        0, overruns:         0, seq_num_broken:   0, pkt_interp:       1764, skew_comp:        0 ms, frm_in_pkt:       2
    Jul 19 14:14:12 10.70.5.205 [DAA]: FXO ONHOOK MONITOR
    Jul 19 14:14:12 10.70.5.205 [DAA]: FXO ONHOOK MONITOR
    Jul 19 14:14:12 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 14:14:12 10.70.5.205 SIP/2.0 200 OK  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-52262f4b;received=10.70.5.205;rport=5060  From: "Cell Phone  " <sip:5555551212@10.10.5.205>;tag=SP168860ab852bfb6c1  To: <sip:36226@10.10.5.205>;tag=as2a5835f9  Call-ID: fefe4615@10.70.5.205  CSeq: 8002 BYE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Content-Length: 0[/size]

OBiSupport

Thanks for the info.

You may log the case that Asterisk cannot receive digits when you call using a Verizon Droid, LG or Samsung cell phone. Send the syslog to us at support@obihai.com, and state your 9-digit OBi number in your email as well. Repeat the same testing scenario is fine.

Asterisk does not seem to accept Inband DTMF signal, so, set to "auto" or "RFC2833" is fine.

RonR

Quote from: otter117 on July 19, 2011, 01:35:30 PM
Setting DTMFmethod to InBand did not help. It actually resulted in Asterisk not "hearing" digits from the iPhone even though they were logged by the Obi110 to my syslog.

Asterisk is notorious for not getting along with InBand DTMF.  Asterisk prefers RFC2833.

otter117

Per your request to open a ticket I sent the log below via e-mail.

Here is my syslog for a call from a Verizon Samsung cell phone.
DTMFmethod = auto, channel gains are all at defaults.

I  pushed the 1 key 10 times and it was only recognized twice.

I spoke with someone who said it might be the cell companies use of the G.729 protocol which can be programmed per switch.
Therefore you might get different results depending upon your carrier and where you dial from.

I have been trying various channel gain values without any improvement.


Jul 19 18:56:07 10.70.5.205 [DAA]: FXO ring on
Jul 19 18:56:09 10.70.5.205
  • DAA CND 07191849,,UNAVAILABLE    ,,,
    Jul 19 18:56:11 10.70.5.205 sendto a0a05cd:5060(952)
    Jul 19 18:56:11 10.70.5.205 INVITE sip:36226@10.10.5.205:5060 SIP/2.0  Call-ID: d4403b3b@10.70.5.205  Content-Length: 400  CSeq: 8001 INVITE  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  Max-Forwards: 70  To: <sip:36226@10.10.5.205>  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-270df828;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384  Contact: <sip:obi110@10.70.5.205:5060>  Expires: 60  Supported: replaces  Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER  Remote-Party-ID: <sip:obi110@10.10.5.205>;party=calling;privacy=off  Content-Type: application/sdp    v=0  o=- 17175 1 IN IP4 10.70.5.205  s=-  c=IN IP4 10.70.5.205  t=0 0  m=audio 10033 RTP/AVP 0 8 18 104 102 103 105 101  a=rtpmap:0 PCMU/8000  a=rtpmap:8 PCMA/8000  a=rtpmap:18 G729/8000  a=rtpmap:104 G726-32/8000  a=rtpmap:102 G726-16/8000  a=rtpmap:103 G726-24/8000  a=rtpmap:105 G726-40/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-15  a=sendrecv  a=ptime:20  a=xg726bitorder:big-endian
    Jul 19 18:56:11 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 18:56:11 10.70.5.205 SIP/2.0 100 Trying  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-270df828;received=10.70.5.205;rport=5060  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  To: <sip:36226@10.10.5.205>  Call-ID: d4403b3b@10.70.5.205  CSeq: 8001 INVITE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Contact: <sip:36226@10.10.5.205>  Content-Length: 0
    Jul 19 18:56:11 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 18:56:11 10.70.5.205 SIP/2.0 200 OK  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-270df828;received=10.70.5.205;rport=5060  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  To: <sip:36226@10.10.5.205>;tag=as4c7df965  Call-ID: d4403b3b@10.70.5.205  CSeq: 8001 INVITE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Contact: <sip:36226@10.10.5.205>  Content-Type: application/sdp  Content-Length: 211    v=0  o=root 15522 15522 IN IP4 10.10.5.205  s=session  c=IN IP4 10.10.5.205  t=0 0  m=audio 10006 RTP/AVP 0 101  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=ptime:20  a=sendrecv
    Jul 19 18:56:11 10.70.5.205 RTP:DtmfTxMtd:1(1),0
    Jul 19 18:56:11 10.70.5.205 sendto a0a05cd:5060(324)
    Jul 19 18:56:11 10.70.5.205 ACK sip:36226@10.10.5.205 SIP/2.0  Call-ID: d4403b3b@10.70.5.205  Content-Length: 0  CSeq: 8001 ACK  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  Max-Forwards: 70  To: <sip:36226@10.10.5.205>;tag=as4c7df965  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-604cd22d;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384
    Jul 19 18:56:11 10.70.5.205 [DAA]: FXO OFFHOOK
    Jul 19 18:56:11 10.70.5.205 [DAA]: detecting disconnect tone, reinit: 1
    Jul 19 18:56:11 10.70.5.205 RTP:Start->a0a05cd:10006(80);0;0;0:0:0;0(15)
    Jul 19 18:56:11 10.70.5.205 [DAA] FXO ring off
    Jul 19 18:56:21 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 175761 ----
    Jul 19 18:56:21 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 18:56:21 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 175821 ms----
    Jul 19 18:56:30 10.70.5.205 [DSP]  ---- S/W DTMF ON (level: 3) : 1, @ 184041 ----
    Jul 19 18:56:30 10.70.5.205 [RTP] DTMF TX (RFC) -- dgt: 1, c_dgt:
    Jul 19 18:56:30 10.70.5.205 [DSP]  ---- S/W DTMF OFF @ 184101 ms----
    Jul 19 18:57:04 10.70.5.205 t_on: 250, t_off: 250, steady_on: 210, steady_off: 290, hits: 5
    Jul 19 18:57:04 10.70.5.205 [DSP]: fxo tone detected: reorder, 3!
    Jul 19 18:57:04 10.70.5.205 [DSP]: fxo (disconnect) tone detected: reorder, 3!
    Jul 19 18:57:04 10.70.5.205
  • DAA Disc.Tone 0 0
    Jul 19 18:57:04 10.70.5.205 sendto a0a05cd:5060(407)
    Jul 19 18:57:04 10.70.5.205 BYE sip:36226@10.10.5.205 SIP/2.0  Call-ID: d4403b3b@10.70.5.205  Content-Length: 0  CSeq: 8002 BYE  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  Max-Forwards: 70  To: <sip:36226@10.10.5.205>;tag=as4c7df965  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-29b40c7a;rport  User-Agent: OBIHAI/OBi110-1.2.1.2384  X-RTP-Stat: PS=2631,OS=450036,PR=2323,OR=399556,PL=0,JI=1,DU=52,EN=G711U,DE=G711U
    Jul 19 18:57:04 10.70.5.205 RTP:Del Channel
    Jul 19 18:57:04 10.70.5.205 [JB] call overall status --       peer:             10.10.5.205:10006, local:            10.70.5.205:10033, pkt_tx:           2631, pkt_rx:           2323, bytes_tx:         450036, bytes_rx:         399556, clk_diff:         497 PPM, pkt_in_jb:        8, pkt_ooo:          0, pkt_lost:         0, pkt_late:         0, pkt_loss_rate:    0 %, pkt_drop_rate:    0 %, jb_len:           190 ms, curr_rcvd_jitter: 1 ms, rcvd_digits:      0, underruns:        0, overruns:         0, seq_num_broken:   0, pkt_interp:       619, skew_comp:        0 ms, frm_in_pkt:       2
    Jul 19 18:57:04 10.70.5.205 [DAA]: FXO ONHOOK MONITOR
    Jul 19 18:57:04 10.70.5.205 [DAA]: FXO ONHOOK MONITOR
    Jul 19 18:57:04 10.70.5.205 RxFrom:a0a05cd:5060
    Jul 19 18:57:04 10.70.5.205 SIP/2.0 200 OK  Via: SIP/2.0/UDP 10.70.5.205:5060;branch=z9hG4bK-29b40c7a;received=10.70.5.205;rport=5060  From: <sip:obi110@10.10.5.205>;tag=SP134752bf759bd403f  To: <sip:36226@10.10.5.205>;tag=as4c7df965  Call-ID: d4403b3b@10.70.5.205  CSeq: 8002 BYE  User-Agent: Asterisk PBX  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO  Supported: replaces  Content-Length: 0

    [/size]

OBiSupport

From your very first post, you stated that OBi 110 DTMF detection works fine with connection to a ILEC POTS line. Correct? if so, OBi has no DTMF problem with regular PSTN network.

The problem appears when you have OBi device connected to an ESI PBX where QWEST is the ILEC.

Since digits go through QWEST VoIP network, we need to diagnose if any distortion is introduced in DTMF signal, particularly when the testing calls are originated from Verizon cell phone. If your PBX connects to a SIP trunk and all the digits (via RFC2833) are received OK, then, local generated digits should work for OBi device. However, if digits are actually sent as inband signal,  speech vocoder might distort DTMF signal badly, e.g., G.729 as you pointed out (even G.711 will have some minor impact), which could be reason why OBi 110 cannot recognize those digits.

Please check with your service provider on whether this is the case, or it would be better that you have a test device to check on DTMF signal that connects to same PBX.

otter117

I got some help from Verizon tech support.
It seems all Verizon phones have a setting for the duration of DTMF tones.
By default, all Verizon phones are set to "normal".
The other setting available is "long".
When I set my Samsung and Android to the "long" setting, the DTMF tones are recognized by the Obi110.
I need to do some further testing/tweaking as it is still not perfect.
I think we will be able to use the Obi110s after all.  ;D

OBiSupport

Thanks for informing us back ... This seems to indicate AT&T iPhones come with a longer DTMF duration by default. Interesting to know this. Good luck to your applications.

otter117

And thank you for your prompt support.

It is going to be an issue where we have to educate the users on changing their phone settings.
Are there any values I can tweak in the OBi110 that will make it recognize the shorter DTMF duration?
I've tried several things like X_UseFixedDurationRFC2833DTMF and DTMFMethod but no luck yet.

OBiSupport

The parameters you refer to, such as X_UseFixedDurationRFC2833DTMF and DTMFMethod, are mainly for transmitting DTMF digits from voice services, and they have no effect on FXO DTMF detection. Actually at this point, DTMF detection on LINE port is not configurable.

However, we think the problem might be that the QWEST voip gateway does not pick up digits properly when you have short DTMF tone. OBi does just fine when there is no QWEST voip network involved, so one thing you might try is to use some other standard DTMF testing device, and connect it to your PBX, and see if signals are OK.

otter117

Thanks,
I have made some progress by setting the Line Port ChannelRxGain to 15
Now, the OBi110 can detect the DTMF with a Droid set to the "normal" DTMF duration setting.
However, it fails on about every 4th call. There may be other factors in the network ( like a bad PRI channel to the PBX ) that is causing this.
I will post any rock solid solution I find.

OBiSupport

This seems to indicate that your VoIP service provider delivers DTMF signal with a somehow weak signal strength. It could be either attenuated, or have a lower signal level on digits regeneration, compare to regular PSTN network. Regular telephone phone generates DTMF signal at -5 dBm ~ - 10 dBm level.