Re: [asterisk-users] continue in dialplan when hang up queue
> Check out the 'c' option to Queue() -- available only in >= 1.6. Hi Alex , I've tested c option, but it just work with h option and H option. DTMF '*' Not on releasing call. Is there another way? thanks! -- Marcus Vinicius-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] continue in dialplan when hang up queue
Hi,
Is there a way to continue dialplan when a call is abandoned from application
queue()?
If the caller is waiting in a queue, and hang up before timeout, I'd like to
execute an application in dialplan.
I've tested "h" exten, but it doesn't work for this.
-- Executing [s@macro-nx-queue:21] Queue("SIP/1019-0c67", "pabx,t") in
new stack
-- Started music on hold, class 'default', on SIP/1019-0c67
-- Called SIP/1021
-- SIP/1021-0c68 is ringing
-- Stopped music on hold on SIP/1019-0c67
== Spawn extension (macro-nx-queue, s, 21) exited non-zero on
'SIP/1019-0c67' in macro 'nx-queue'
== Spawn extension (from-inside-redir, *5000, 1) exited non-zero on
'SIP/1019-0c67'
I'D LIKE TO CONTINUE DIAL PLAN HERE
-- Executing [h@from-inside-redir:1] Hangup("SIP/1019-0c67", "") in new
stack
== Spawn extension (from-inside-redir, h, 1) exited non-zero on
'SIP/1019-0c67'
== Extension Changed 1019[from-inside] new state Idle for Notify User 1033
Thanks a lot!
--
Marcus Vinícius
--
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[asterisk-users] Res: digits in chan_dahdi
Hello, thanks for the reply. I tried relaxdtmf = yes but has not worked. If I type very slowly digits are recognized normally. But if I dial a number and enter the redial button, the digits are recognized in the asterisk. It appears that: [Sep 21 19:20:24] DEBUG [4751] chan_dahdi.c: waitfordigit returned <0 ... tks Marcus Vinicius De: Richard Kenner Para: asterisk-users@lists.digium.com Enviadas: Terça-feira, 21 de Setembro de 2010 18:48:54 Assunto: Re: [asterisk-users] digits in chan_dahdi > I dial 12345678, but only '16 'is received by the asterisk. You may want to try relaxdtmf=yes in chan_dahdi.conf. That fixed a similar problem for me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] digits in chan_dahdi
Hello I use Asterisk with FXS extensions in chan_dahdi and I'm having trouble detecting the digits in dahdi. I dial 12345678, but only '16 'is received by the asterisk. The following appears in the logs: [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end '1 'received on DAHDI/10-1, duration 0 ms [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end accepted without begin '1 'on DAHDI/10-1 [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end passthrough '1 'on DAHDI/10-1 [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end '6 'received on DAHDI/10-1, duration 0 ms [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end accepted without begin '6 'on DAHDI/10-1 [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end passthrough '6 'on DAHDI/10-1 [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: gotoif [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: SetMusicOnHold [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Goto [Sep 21 18:11:48] DEBUG [8536] chan_dahdi.c: Took DAHDI/10-1 off hook I use the headset Zox TS19. I tried changing the value of toneduration = 100 but did not work. Anyone know how I can solve this problem? thank you very much. Marcus Vinícius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canary_thread
People, Anybody knows what mean this message in my CLI: [Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) mediagw*CLI> Asterisk: 1.6.2.6 tks Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] # as dial key - chan_dahdi
Hi, Can I set up '#' as dial key using the extensions fxs? I use chan_dahdi, and a TDM400P card. I'm testing and, nothing happens when I press #. thanks. -- Marcus Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pickupexten on chan_dahdi
Hi,
I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here
my settings:
chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
facilityenable=yes
rxwink=300 ; Atlas seems to use long (250ms) winks
; where the ring cadence is changed *after* the
callerid spill.
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=0
pickupgroup=0
immediate=no
accountcode=0003.002
callgroup=0
pickupgroup=0
callerid=231
channel=>27
callgroup=0
pickupgroup=0
callerid=232
channel=>28
; interfaces FXO (linha)
context=default
signalling=fxs_ks
group=2
busydetect=yes
callerid=asreceived
channel=>29-36
features.conf
[general]
parkext => 600 ; What extension to dial to park
parkpos => 601-620 ; What extensions to park calls on. These needs to be
; numeric, as Asterisk starts from the start position
; and increments with one for the next parked
call.
context => parkedcalls ; Which context parked calls are in
parkingtime => 180 ; Number of seconds a call can be parked for
; (default is 45 seconds)
transferdigittimeout => 3 ; Number of seconds to wait between digits when
transfering a call
courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' free parking space.
; Defaults to 'first' available
pickupexten = 5 ; Configure the pickup extension. Default is *8
featuredigittimeout = 700 ; Max time (ms) between digits for
; feature activation. Default is 500
When I dial 5 from an analog extension, dial it 5 on the dial plan and does not
capture the call:
-- Executing [...@from-inside:1] Macro("DAHDI/9-1", "nx-set-variables") in
new stack
-- Executing [...@macro-nx-set-variables:1] Set("DAHDI/9-1", "tenant=") in
new stack
-- Executing [...@macro-nx-set-variables:2] Set("DAHDI/9-1",
"CDR(userfield)=") in new stack
Anybody know what might be happening?
Asterisk: 1.4.26
Dahdi: 2.2.0.2
thanks.
--
Marcus
Veja quais são os assuntos do momento no Yahoo! +Buscados
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[asterisk-users] interdigit timeout chan_dahdi
Hello, I have an extension into an analog FXS interface. When taking the unit off the hook and dial any number of digits, it takes about 4 seconds for these digits are passed to the dial plan. Anybody know if this time can be customized? /etc/dahdi/system.conf echocanceller=mg2,1-36 dynamic=eth,eth0/00:18:43:0b:00:46,36,1 fxoks=1-36 /etc/asterisk/chan_dahdi.conf [channels] context=default switchtype=national ;signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks ; where the ring cadence is changed *after* the callerid spill. usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ; interfaces FXS (ramal) context=from-inside signalling=fxo_ks group=1 callerid=226 channel=>1 callerid=200 channel=>2 thanks -- Marcus Vinicius Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with T38 Fax
Hi, I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk rejects the t38. Anybody know if is possible to transmit t38 fax with Asterisk 1.4? following settings: --- sip.conf --- [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw context=from-outside t38pt_udptl=yes [operator] qualify=no nat=yes host=189.160.126.201 dtmfmode=rfc2833 context=from-outside type=friend canreinvite=yes t38pt_udptl=yes ;t38pt_rtp=no ;t38pt_tcp=no disallow=all allow=ulaw allow=alaw --- channels/chan_sip.c --- static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600; --- logs --- logs [Nov 13 10:21:11] VERBOSE[25087] logger.c: <--- SIP read from 189.160.126.210:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060 Call-ID: 21cdaea43523056c3a09c45b13c9a...@189.6.70.47 From: "Teste";tag=as41b028c6 To: ;tag=66359f37 CSeq: 102 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Contact: Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 31955175 31955175 IN IP4 189.160.126.210 s=Sip Call c=IN IP4 189.160.126.210 t=0 0 m=audio 13474 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-> [Nov 13 10:21:11] VERBOSE[25087] logger.c: --- (10 headers 10 lines) --- [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 0 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 8 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 101 [Nov 13 10:21:11] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL [Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMU for ID 0 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMA for ID 8 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format telephone-event for ID 101 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 13 10:21:11] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:11] VERBOSE[13464] logger.c: -- SIP/ctbc-08345a10 is making progress passing it to IAX2/nmg010-to-nmg005-trunk1-2748 <--- SIP read from 189.160.126.210:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060 Call-ID: 21cdaea43523056c3a09c45b13c9a...@189.6.70.47 From: "Teste";tag=as41b028c6 To: ;tag=66359f37 CSeq: 102 INVITE Contact: Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 31955175 31955176 IN IP4 189.160.126.210 s=Sip Call c=IN IP4 189.160.126.210 t=0 0 m=audio 13474 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-> [Nov 13 10:21:15] VERBOSE[25087] logger.c: --- (9 headers 10 lines) --- [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 0 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 8 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 101 [Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL [Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMU for ID 0 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMA for ID 8 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format telephone-event for ID 101 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 13 10:21:15] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:15] VERBOSE[25087] logger.c: list_route: hop: [Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Strict routing enforced for session 21cdaea43523056c3a09c45b13c9a...@189.6.70.47 [Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: Parsing for address/port to send to [Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: set destination to 189.160.126.210, port 5060 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Transmitting (NAT) to 189.160.126.210:5060: ACK sip:0411331644...@189.160.126.210:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6618dc53;rport From: "Teste"
[asterisk-users] Res: Asterisk core dumps files
hi, the -g option is right. make sure that the system allows core files (ulimit -a). Regards -- Marcus De: Gustavo A Gonzalez Para: asterisk-users@lists.digium.com Enviadas: Quinta-feira, 30 de Julho de 2009 11:17:50 Assunto: Re: [asterisk-users] Asterisk core dumps files Thanks Tzafrir for your answer. Because I had some problems running safe_asterisk script to restart asterisk automatically in our callcenter , I’ve developed a simple script that runs from a schedule task and check if asterisk is running each minute. This is not the best solution yet but it works properly when asterisk shutdown. However it not let asterisk generate core dumps files. Is there an error in this script or what I have to change to get core dumps files from this script. #!/bin/sh # #Script para levantar el asterisk automaticamente #programado por WL echo “Checking if asterisk is running” a=`pidof asterisk` if [ "$a" != "" ]; then echo "Everything is OK, Asterisk is UP and running"; else echo "Asterisk Error: NOT RUNNING trying to restart it in 5 attempts!!!"; for ((i=1; i<=5; i+=1)); do /usr/sbin/asterisk -g b=`pidof asterisk` if [ "$b" != "" ]; then exit fi done fi G.A.G. Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polarity Reversal Incorrect
Hi, I have a FXO line in TDM410P card. Over some calls, after a few minutes of conversation, a busy tone occurs on the call. The call remains up, the two sides of the call is heard normally, but remains a busy tone in the middle of the conversation. When this occurs in the log appears the following: Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Got event Polarity Reversal(17) on channel 2 (index 0) Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Ignoring Polarity switch to IDLE on channel 2, state 6 Jul 10 13:22:45 DEBUG[11688] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 2, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= 1706049808 Anybody know how to fix this problem? I tried to add relaxdtmf=yes in zapata.conf but the problem persisted. zapata.conf [channels] context=default rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes restrictcid=no callwaitingcallerid=yes threewaycalling=yes transfer=no cancallforward=no callreturn=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=-2.0 group=1 callgroup=1 pickupgroup=1 faxdetect=no immediate=no musiconhold=default echocancel=yes relaxdtmf=yes context=default busydetect=no context=default signalling=fxs_ks group=2 channel=>1-3 Thank you. -- Marcus Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
