DTMF settings

Asterisk Asterisk Support
1

does anyone know of any DTMF settings within asterisk?

i am having problems where i can dial certain extensions but not others.
ie can dial ext 429 but not 435.

when 435 is dialled the call rings once and then is dropped… in verbose mode, everythign connects the same for both calls.

any ideas or similar problems?

2

what technology are you using to connect ? if you’re using a SIP or IAX device, then Asterisk isn’t processing DTMF to dial. you’ll need to post more info.

3

i am using SIP clients.

i have a general extensions.conf file that should handle all 400 extensions the same… and yet some work and others dont. At one point i had everything working fine, i didn’t modify anything and now its inconsistent.

[chan_one_out]
exten => _4XX,1,Dial(Zap/1/${EXTEN},20)

4

and the CLI output for a failed call ?

5

in verbose mode this is the output for a call on the 435 ext that doesnt work.

*CLI> – Executing Dial(“SIP/3000-03e0”, “Zap/1/435|20”) in new stack
– Called 1/435
– Zap/1-1 answered SIP/3000-03e0
– Hungup ‘Zap/1-1’
== Spawn extension (chan_one_out, 435, 1) exited non-zero on ‘SIP/3000-03e0’

and this is the output for a call on ext 429 that does work

-- Executing Dial("SIP/3000-195a", "Zap/1/429|20") in new stack
-- Called 1/429
-- Zap/1-1 answered SIP/3000-195a
-- Hungup 'Zap/1-1'

== Spawn extension (chan_one_out, 429, 1) exited non-zero on ‘SIP/3000-195a’

6

what are you connecting to ? and what does your zapata.conf look like ?

7

what i am trying to connect four different SIP clients to our existing PBX, its an NEC system.

this is my zapata.conf file-

[code][channels]
signalling=fxs_ks
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
busydetect=yes
busycount=3
rxgain=10.0
txgain=10.0
echocancel=yes
echotraining=yes
immediate=no

context=chan_one_in
channel => 1
context=chan_two_in
channel => 2
context=chan_three_in
channel => 3
context=chan_four_in
channel => 4[/code]

this is a piece of my extensions.conf file to give you an idea of what im trying to do.

[code][chan_one_in]
exten => s,1,Dial(SIP/3000,20)
exten => h,1,Hangup()

[chan_one_out]
exten => _4XX,1,Dial(Zap/1/w${EXTEN},20)
exten => _5XX,1,Dial(Zap/1/w${EXTEN},20)
exten => _6XX,1,Dial(Zap/1/w${EXTEN},20)
exten => _9NXXNXXXXXX,1,Dial(Zap/1/w${EXTEN},20)
exten => _91NXXNXXXXXX,1,Dial(Zap/1/w${EXTEN},20)

[chan_two_in]
exten => s,1,Dial(SIP/3001,20)
exten => h,1,Hangup()

[chan_two_out]
exten => _4XX,1,Dial(Zap/2/${EXTEN},20)
exten => _5XX,1,Dial(Zap/2/${EXTEN},20)
exten => _6XX,1,Dial(Zap/2/${EXTEN},20)
exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN},20)
exten => _91NXXNXXXXXX,1,Dial(Zap/2/${EXTEN},20)[/code]

8

i’m assuming that in sip.conf, you have individual context’s set for each device, like so:

[sip1]
context=chan_one_out
…

you might also try adding a ‘debug’ to the logger.conf file for the console entry (console => notice,warning,error,debug), doing a ‘logger reload’ on the CLI, and trying this again. we might need some debug info to proceed - i’m wondering if it might be a transcoding issue.

in any case, set your verbose and debug to 10 and try another call and post the results.

9

yes that is correct this is one user in my sip.conf

[3000] type=friend host=dynamic username=3000 careinvite=no defaultip=xxx.xxx.xxx.xxx dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=chan_one_out

ok i ran a call to ext 435 again and this is what i got

*CLI> Jul 10 16:06:20 DEBUG[14866]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Jul 10 16:06:20 DEBUG[14866]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 3000 Jul 10 16:06:20 DEBUG[14866]: chan_sip.c:6137 build_route: build_route: Contact hop: <sip:3000@192.168.10.108:5060> -- Executing Dial("SIP/3000-f547", "Zap/1/435") in new stack Jul 10 16:06:20 DEBUG[14881]: dsp.c:1619 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 Jul 10 16:06:20 DEBUG[14881]: chan_zap.c:1916 zt_call: Dialing '435' Jul 10 16:06:20 DEBUG[14881]: chan_zap.c:1988 zt_call: Deferring dialing... -- Called 1/435 Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:3538 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:3538 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:1405 zt_enable_ec: Enabled echo cancellation on channel 1 Jul 10 16:06:21 DEBUG[14881]: chan_zap.c:1421 zt_train_ec: Engaged echo training on channel 1 Jul 10 16:06:23 DEBUG[14881]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:06:23 DEBUG[14881]: chan_zap.c:3538 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Jul 10 16:06:23 DEBUG[14881]: chan_zap.c:1385 zt_enable_ec: Echo cancellation already on -- Zap/1-1 answered SIP/3000-f547 Jul 10 16:06:23 DEBUG[14859]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/3000-f547' Jul 10 16:06:24 DEBUG[14866]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '85119536684807542@192.168.10.108' of Response 1755956448: Match Found Jul 10 16:06:31 DEBUG[14881]: dsp.c:1294 ast_dsp_busydetect: ast_dsp_busydetect detected busy, avgtone: 280, avgsilence 225 Jul 10 16:06:31 DEBUG[14881]: dsp.c:1436 ast_dsp_process: Requesting Hangup because the busy tone was detected on channel Zap/1-1 Jul 10 16:06:31 DEBUG[14881]: channel.c:3287 ast_generic_bridge: Got a FRAME_CONTROL (5) frame on channel Zap/1-1 Jul 10 16:06:31 DEBUG[14881]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels SIP/3000-f547 and Zap/1-1 Jul 10 16:06:31 DEBUG[14881]: chan_zap.c:2344 zt_hangup: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jul 10 16:06:31 DEBUG[14881]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 Jul 10 16:06:31 DEBUG[14881]: chan_zap.c:2784 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jul 10 16:06:31 DEBUG[14881]: chan_zap.c:1374 update_conf: Updated conferencingon 1, with 0 conference users -- Hungup 'Zap/1-1' Jul 10 16:06:31 DEBUG[14881]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. == Spawn extension (chan_one_out, 435, 1) exited non-zero on 'SIP/3000-f547' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '3000' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '3000' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '435' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'chan_one_out' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/3000-f547' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Zap/1-1' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Zap/1/435' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:06:20' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:06:23' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:06:31' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '11' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '8' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1152561980.0' Jul 10 16:06:31 DEBUG[14881]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 10 16:06:31 DEBUG[14881]: chan_sip.c:2426 sip_hangup: update_call_counter(3000) - decrement call limit counter Jul 10 16:06:32 DEBUG[14866]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '85119536684807542@192.168.10.108' of Request 102: Match Found

10

i am not very well versed in the analog side of things, but it appears as though the asterisk box is receiving a hangup signal from the NEC system - can you verify that your signalling is correct for the trunks that tie the two systems together? that would be my first guess, but i admit that it is just a guess.

11

this is the log for a call that connects on a different extension

*CLI> Jul 10 16:36:40 DEBUG[18754]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Jul 10 16:36:40 DEBUG[18754]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 3000 Jul 10 16:36:40 DEBUG[18754]: chan_sip.c:6137 build_route: build_route: Contact hop: <sip:3000@192.168.10.108:5060> -- Executing Dial("SIP/3000-defd", "Zap/1/469") in new stack Jul 10 16:36:40 DEBUG[18782]: dsp.c:1619 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 Jul 10 16:36:40 DEBUG[18782]: chan_zap.c:1916 zt_call: Dialing '469' Jul 10 16:36:40 DEBUG[18782]: chan_zap.c:1988 zt_call: Deferring dialing... -- Called 1/469 Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:3538 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:3538 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:1405 zt_enable_ec: Enabled echo cancellation on channel 1 Jul 10 16:36:41 DEBUG[18782]: chan_zap.c:1421 zt_train_ec: Engaged echo training on channel 1 Jul 10 16:36:43 DEBUG[18782]: chan_zap.c:4350 __zt_exception: Exception on 16, channel 1 Jul 10 16:36:43 DEBUG[18782]: chan_zap.c:3538 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Jul 10 16:36:43 DEBUG[18782]: chan_zap.c:1385 zt_enable_ec: Echo cancellation already on -- Zap/1-1 answered SIP/3000-defd Jul 10 16:36:43 DEBUG[18747]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/3000-defd' Jul 10 16:36:43 DEBUG[18754]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '356440103566611854@192.168.10.108' of Response 1683461313: Match Found Jul 10 16:36:47 DEBUG[18782]: channel.c:3275 ast_generic_bridge: Didn't get a frame from channel: SIP/3000-defd Jul 10 16:36:47 DEBUG[18782]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels SIP/3000-defd and Zap/1-1 Jul 10 16:36:47 DEBUG[18782]: chan_zap.c:2344 zt_hangup: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jul 10 16:36:47 DEBUG[18782]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 Jul 10 16:36:47 DEBUG[18782]: chan_zap.c:2784 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jul 10 16:36:47 DEBUG[18782]: chan_zap.c:1374 update_conf: Updated conferencingon 1, with 0 conference users -- Hungup 'Zap/1-1' Jul 10 16:36:47 DEBUG[18782]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. == Spawn extension (chan_one_out, 469, 1) exited non-zero on 'SIP/3000-defd' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '3000' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '3000' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '469' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'chan_one_out' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/3000-defd' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Zap/1-1' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Zap/1/469' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:36:40' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:36:43' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-10 16:36:47' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '7' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '4' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1152563800.0' Jul 10 16:36:47 DEBUG[18782]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 10 16:36:47 DEBUG[18782]: chan_sip.c:2426 sip_hangup: update_call_counter(3000) - decrement call limit counter

12

it seems that asterisk doesnt recognize all the numbers being dialled is there a way to slow down the dialing on the asterisk system?

13

lets see…

dialing to Zap/1/469:

[quote]…
Jul 10 16:36:43 chan_sip.c:1401 __sip_ack: Stopping retransmission on ‘356440103566611854@192.168.10.108’ of Response 1683461313: Match Found
Jul 10 16:36:47 channel.c:3275 ast_generic_bridge: Didn’t get a frame from channel: SIP/3000-defd
…[/quote] looks like SIP/3000 made a hangup

dialing to Zap/1/435:

[quote]…
Jul 10 16:06:24 chan_sip.c:1401 __sip_ack: Stopping retransmission on ‘85119536684807542@192.168.10.108’ of Response 1755956448: Match Found
Jul 10 16:06:31 dsp.c:1294 ast_dsp_busydetect: ast_dsp_busydetect detected busy, avgtone: 280, avgsilence 225
Jul 10 16:06:31 dsp.c:1436 ast_dsp_process: Requesting Hangup because the busy tone was detected on channel Zap/1-1
… [/quote]looks like 435 is in busy state

try to check ${DIALSTATUS} from Dial
example i have found on the net:
exten => s,1,Dial(IAX2/400,20,t)
exten => s,n,NoOP(Dial Status: ${DIALSTATUS})

-F