[asterisk-users] Playback on h exten
Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a
agent hangup.
I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g
but every time I try to play something I got:
-- Executing [301@from-test:1] Dial(SIP/300-0045, SIP/301,60,rjtTg)
in new stack
-- Called SIP/301
-- SIP/301-0046 is ringing
-- SIP/301-0046 answered SIP/300-0045
-- Auto fallthrough, channel 'SIP/300-0045' status is 'ANSWER'
-- Executing [h@from-test:1] Goto(SIP/300-0045, play,s,1) in new
stack
-- Goto (play,s,1)
-- Executing [s@play:1] NoOp(SIP/300-0045, play) in new stack
-- Executing [s@play:2] SayDigits(SIP/300-0045, 123579) in new stack
[Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback: Failed to
write frame
-- SIP/300-0045 Playing 'digits/1.ulaw' (language 'en')
== Spawn extension (play, s, 2) exited non-zero on 'SIP/300-0045'
This is my dialplan:
[from-test]
exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
exten = h,1,Goto(play,s,1)
[play]
exten = s,1,Noop(play)
exten = s,2,Saydigits(123579)
Anyone can help me?
Thanks
Enrico.
--
--
Pasqualotto Enrico
cell. +39 3473292620
skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto
http://www.netspin.it :: e.pasqualo...@netspin.it
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Re: [asterisk-users] Playback on h exten
Yes, correct now it works for Dial.
I think is the same with c option on Queue, do you think there's a way to do
it on h exten?
My goal is to inject my dialplan on hangup macro.
Enrico.
- Messaggio originale -
If you choose to go with the Dial command and use the g option, you
have not to use the h extension, but just provide a next priority.
Your dialplan has to be:
[from-test]
exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
exten = _X.,2,Goto(play,s,1)
[play]
exten = s,1,Noop(play)
exten = s,2,Saydigits(123579)
Leandro
--
--
Pasqualotto Enrico
cell. +39 3473292620
skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto
http://www.netspin.it :: e.pasqualo...@netspin.it
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Re: [asterisk-users] Logging Asterisk console
On Tue, 2009-04-07 at 15:21 +0200, Marco Sambo wrote: Hi Enrico, I do that by modifying logger.conf [logfiles] logpro = notice,warning,error,debug,verbose and modifying asterisk.conf [directories] astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk [options] verbose = 3 and so I find into /var/log/asterisk the logpro file with the output of CLI (verbose) and notice, warning, error, debug message of Asterisk. Yes, but with this log I can't see the list of application called. Enrico. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Logging Asterisk console
Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with asterisk -rvvv. I need it in debugging purpose for tracking some bug. Thanks Enrico. smime.p7s Description: S/MIME cryptographic signature ___ -- 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 BLF to Cisco CME
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[asterisk-users] app transfer problem
I all, I'm trying to transfer a iax2 channel trought dialplan app transfer to another extensions (IAX). The variable TRANSFERSTATUS report SUCCESS but the call isn't trasfered. I haven't other information, in console I see only hangup of a channel. My scenario is 3 asterisk box connected with iax trunk, I talk from BOX1 to BOX2 and I want to transfer user on BOX2 to another user on BOX3. After transfer command is executed on BOX1 (return SUCCESS) on other BOX I don't see nothing. Only hangup. There are some way to debug the transfer? Thanks Pasqu. ___ -- 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] Fax issue over cisco gateway
Hi all, I'm trying to send fax from Hylafax to a remote fax machine through Asterisk and cisco 2801 as E1 gateway. This is my architecture: sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card For incoming fax I don't have any problem, but I'm not able to send fax out of 2801. My router conf: dial-peer voice 1 pots destination-pattern .T fax rate disable port 0/2/0:15 ! dial-peer voice 3 pots incoming called-number 53T fax rate disable direct-inward-dial forward-digits all ! ## In asterisk console I see a lot of RTP packets lost: RTP-stats-003*CLI * Our Receiver: SSRC: 642188040 Received packets: 17463 Lost packets: 19686 Jitter:0. Transit: 0. RR-count: 0 * Our Sender: SSRC: 1469234407 Sent packets: 27926 Lost packets: 0 Jitter:0 SR-count: 112 RTT: 0.00 Anyone have idea of this problem? The packet lost quantity is normal? Thanks Enrico. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] Asterisk Realtime with MySQL Registration Failed
On Mon, 2008-08-04 at 16:48 +0300, Abid Saleem wrote: May be. If somebody has experience this problem before, then only he/she can guide about this. I am not sure whats going on. Abid Saleem Try to set debug verbose option in logger.conf, then check all query from asterisk to mysql for see what goes wrong. Pasqu. smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] wait pickup
Hi all, One question I have set in the extensions.conf of my asterisk that all incoming call go in the wait application because I need to not connect the caller but remain in the ringing state. After that the call is on the wait exten for a N second I need from other sip phone to pickup this call. There is a way to pickup a call arrived from IAX to an exten wait(999)? I see that the problem is the channel state, my channel in wait is in LINE IS RING but the pickup appl search for channel in REMOTE RINGING. Anyone have solution for this problem? TIA Pasqu... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] wait pickup
On Thu, 2008-07-03 at 09:31 -0500, Eric ManxPower Wieling wrote: chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8). Wow! It's a very nice problem And for redirect a call in wait state to a sip phone? Without pickup ... Channelredirect don't work with ringing channel for me. Thanks Pasqu. smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Conference Hangup
Hi all, I have a question on asterisk conference. Now I use appl Meetme with option A x because when a marked person hangup I want to close all the conference. But what I have to do if I want two marked person and kill the conference when one of two hangup? Is possible? Thanks. Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Realtime context
Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include = parkedcalls switch = Realtime/@ [fromiax] switch = Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch = Realtime/@ but I have about 50 asterisk that read one database, now if I want to change/add a context I have to change 50 extensions.conf file :( Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --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 translator issue?
Hi all, I have a network with some asterisk in trunk with IAX2 and some SIP/ZAP phone connect to this *. In every call I need to use only alaw codec so in all conf file I have set disallow=all and allow=alaw. I try also to make some tuning of my environment removing unused codec and application. If I remove the codec_ulaw.so when I try to call I see this: [Oct 5 12:15:33] WARNING[16637]: chan_iax2.c:8021 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/CTM1-283 -- Hungup 'IAX2/CTM1-283' [Oct 5 12:15:33] WARNING[16637]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) instead I use only alaw. Infact if I keep che codec_ulaw.so and during a call watch the used codec all are alaw. Anyone can explain me where is the problem? P.S. for me is not a problem to keet one file but is interesting to know who want to translate who. Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Park problem on IAX2 channel
Hi all, I have 2 asterisk box connected with IAX trunk. One box have connected a SIP phone and the second have a TDM card with one analog phone. When from SIP phone I try to park the call from analog phone with #700 the call is correctly parked but in the second asterisk I see this log: -- Executing Dial(Zap/2-1, IAX2/CTM1/STI1|30|rjtT) -- Called CTM1/STI1 -- Call accepted by 172.16.4.1 (format alaw) -- Format for call is alaw -- IAX2/CTM1-2 answered Zap/2-1 -- Started music on hold, class 'default', on IAX2/CTM1-2 -- Zap/2-1 Playing 'pbx-transfer' (language 'en') -- Unable to find extension '77' in context 'from-internal' -- Zap/2-1 Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on IAX2/CTM1-2 The line: -- Unable to find extension '77' in context 'from-internal' appears also with '#', '#7', '', '0'... It seems that the dtmf came across the iax channel and arrive to other asterisk. The are a way to block this dtmf across the IAX trunk? Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Dial and option G
Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten = ,1,Dial(IAX2/DIP02/||G(fromiax^^1) [fromiax] exten = ,1,MeetMe(,qdxAa) exten = ,2,MeetMe(,qdx) exten = ,3,Dial(other-user,,G(from-iax,,4)) exten = ,4,MeetMe(,qdx) but not work. Any suggestion? Thanks Enrico -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --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] mISDN problems
Hi all, we're buildin an Asterisk box based on an Intel IXP425 board. The board uses a Beronet BN2S0 ISDN card, mISDN 1.1.4 and asterisk 1.4.2. hfc_multi has been patched to compile under big endian cpu, and so also capi kernel files. All the modules seem to load correctly (configuration was made with misdn-init config), but when starting cha_misdn, asterisk outputs the following lines: P[ 1] Restarting this port. P[ 1] Stack:0x174f10 P[ 1] empty_chan_in_stack: 1 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] empty_chan_in_stack: 2 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] empty_chan_in_stack: 3 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] L1: PH L1Link Up! P[ 0] MGMT: SSTATUS: L1_ACTIVATED P[ 1] % GOT L2 DeActivate Info. P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! P[ 1] % GOT L2 Activate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! This error is generated by a function into asterisk package in file channels/misdn/isdn_lib.c misdn-init start output: dip01:/mnt/externfs/beronet/install-misdn-mqueue/mISDN-1_1_4# misdn-init start - Loading module(s) for your misdn-cards: - /sbin/modprobe --ignore-install hfcmulti type=0x1 protocol=0x12,0x22 layermask=0x3,0xf poll=128 debug=0x88 /sbin/modprobe mISDN_dsp debug=0x0 options=0 poll=160 dtmfthreshold=100 dmesg related output: Modular ISDN Stack core version (1_1_4) revision ($Revision: 1.40 $) mISDNd: kernel daemon started (current:c2c2bac0) ISDN L1 driver version 1.20 mISDNd: test event done ISDN L2 driver version 1.32 mISDN: DSS1 Rev. 1.47 mISDN Capi 2.0 driver file version 1.21 mISDN: HFC-multi driver Rev. 1.68 HFC-multi: card manufacturer: 'Cologne Chip AG' card name: 'HFC-2S Beronet Card' clock: double PCI: enabling device :00:05.0 ( - 0003) HFC-2S#1: defined at IOBASE 0x1000 IRQ 28 HZ 100 leds-type 3 HFC_multi: resetting HFC with chip ID=0xc revision=1 hfcpci_probe: DIPs(0x9f) jumpers(0x1) HFC_manager: channel 2 (0..31) data c30d prim f1681 arg HFC_manager: MGR_REGLAYER HFC_manager: channel 2 (0..31) data c30d prim f1482 arg HFC_manager: MGR_SETSTACK HFC_manager: channel 2 (0..31) data c30d prim f4182 arg HFC_manager: channel 6 (0..31) data c63b2800 prim f1681 arg HFC_manager: MGR_REGLAYER HFC_manager: channel 6 (0..31) data c63b2800 prim f1a82 arg cb150e50 HFC_manager: MGR_***STPARA HFC_manager: channel 6 (0..31) data c63b2800 prim f1a82 arg cb150e50 HFC_manager: MGR_***STPARA HFC_manager: channel 6 (0..31) data c63b2800 prim f1482 arg HFC_manager: MGR_SETSTACK 1 devices registered HFC_manager: channel 6 (0..31) data c63b2800 prim f4182 arg mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) mISDN_dsp: DSP clocks every 160 samples. This equals 2 jiffies. The only output that we see when interacting with a phone connected to a PBX is a string like 0x64 0x7f 0x01 but it seems more related to layer1. Once we have seen an error from mISDN_read, coded 22, but never have been able to reproduce it. The same configuration on x86 work perfectly. Any idea? Regards smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Beronet card - issue?
Hi all, I have a problem with my beronet card with 2 isdn. I think drivers and Asterisk are ok but the red led on the card always blinking. The card is connected with PBX. I post some conf: [EMAIL PROTECTED] ~]# misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - childcnt: 2 mISDN_close: fid(3) isize(131072) inbuf(0x9eff060) irp(0x9eff060) iend(0x9eff060) [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# cat /etc/misdn-init.conf card=1,0x4 te_ptmp=1,2 poll=127 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=5 [EMAIL PROTECTED] ~]# cat /etc/asterisk/misdn.conf [general] debug = 0 method=standard bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=it musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=no reject_cause=16 need_more_infos=no nttimeout=no method=standard dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no nodialtone=no callgroup=1 pickupgroup=1 presentation=-1 screen=-1 echocancel=yes jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no [isdn] ports=1 context=from-pstn msns=* This is the first time that I configure this type of card Link of some good docs is ok too. :) Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
Rob Schall wrote: One easy way to get close to this affect: Create a group dialDial(SIP/1000SIP/1001) then have a dynamic meetme room generating extension. This way, you can put them on hold for a brief second, dial that extension, create a room, then transfer them into it. This keeps the number of conference rooms to a min, while letting you create them on the fly for when you need more than 3 people on a call. Rob Thanks Rob, another way (I think): I make a standard 2 way call (2000 to 2001), if other user (2002) call 2000 or 2001 and the DIALSTATUS is busy using channelredirect I put the three user in one conference. I think this is MY solution... Now I try! -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using meetme like call
hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because for the users is the simplest (I think). The problem is that when user call one extension that isn't available or not responding the first user remain in the room for all work day. :( There's a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using meetme like call
hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because for the users is the simplest (I think). The problem is that when user call one extension that isn't available or not responding the first user remain in the room for all work day. :( There's a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Realtime - show peers works?
hi all, I'm trying to set up some iax2 trunks in Realtime architecture
with the same backend.
All work better (make call, receive etc etc) but when I do iax2 show
peers some asterisk don't show anything and other show the iax2 peers
but with status unknow.
Name/UsernameHost Mask Port
Status
ctm1/trixbox 10.0.0.131 (S) 255.255.255.255 4569
UNKNOWN
I have set in iax.conf rtcachefriends=yes but the status not change.
There are anyone with this situation that iax2 show peers work?
One of my mysql records:
INSERT INTO `iax_buddies` (`name`, `username`, `type`, `secret`,
`md5secret`, `dbsecret`, `notransfer`, `inkeys`, `outkey`, `auth`,
`accountcode`, `amaflags`, `callerid`, `context`, `defaultip`, `host`,
`language`, `mailbox`, `deny`, `permit`, `qualify`, `disallow`, `allow`,
`ipaddr`, `port`, `regseconds`) VALUES
('ctm2', 'trixbox', 'friend', 'X', NULL, NULL, NULL, NULL, NULL,
NULL, NULL, NULL, NULL, 'fromiax', NULL, '10.0.0.254', NULL, NULL, NULL,
NULL, 'yes', 'all', 'alaw', NULL, 4569, 0);
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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[asterisk-users] Asterisk CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323.
Cisco conf:
dial-peer voice 8 voip
destination-pattern 2...
session target ipv4:asterisk ip
codec g711alaw
no vad
h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=from-internal
extension.conf
[from-internal]
exten = _1XXX,1,Dial(SIP/${EXTEN}@cme ip)
exten = 2000,1,Dial(SIP/2000)
I'm able from Asterisk to call ip phone connected to cme but from cme to
asterisk the phones ring but go in hangup immediatly.
My debug:
---
localhosAnswering call ip$192.168.99.2:53716/21
localhos-- Transmitting RFC2833 on payload 101
localhos-- Received Facility message...
localhos-- Received Facility message...
localhos-- Inbound RFC2833 on payload 101
localhos-- Received RELEASE COMPLETE message...
localhos-- ClearCall: Request to clear call with token
ip$192.168.99.2:53716/21, cause 22
localhos-- Sending RELEASE COMPLETE
localhost*CLI channelsOpen = 1
channelsOpen = 0
localhos-- ClearCall: Request to clear call with token
ip$192.168.99.2:53716/21, cause 7
Scheduling destruction of call
'[EMAIL PROTECTED]' in 32000 ms
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port
to send to
set_destination: set destination to 192.168.99.122, port 5060
Reliably Transmitting (no NAT) to 192.168.99.122:5060:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: 1003 sip:[EMAIL PROTECTED];tag=as769a2c55
To: sip:[EMAIL PROTECTED]:5060;tag=1473512925
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
-- Call with Enrico [192.168.99.2] completed (22)
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhost*CLI
-- SIP read from 192.168.99.122:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: 1003 sip:[EMAIL PROTECTED];tag=as769a2c55
To: sip:[EMAIL PROTECTED]:5060;tag=1473512925
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
Server: X-Lite release 1105d
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhos== H.323 Connection deleted.
I don't understand why the call goes down only from cisco to
asterisk any ideas?
Thanks Enrico
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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[asterisk-users] *****SPAMZ***** Conference Page question
Spam detection software, running on the system placebo, has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
[EMAIL PROTECTED] for details.
Content preview: Hi. I'm currently setting up a particular conference: 3
members (a,b,c), a can speak with b and c, b and c can speak only with a
and not between them. I found my possible solution with paging/intercom
using option d (full-duplex), but I need to make ringing the phone in
intercom. Now I set auto-answer=6 but after first ring the phone hangup
the call. There is a way to using page/intercom with normal ring and not
with auto-answer? [...]
Content analysis details: (4.1 points, 4.0 required)
pts rule name description
-- --
5.0 BOTNET Relay might be a spambot or virusbot
[botnet0.7,ip=82.184.107.109,hostname=host109-107-static.184-82-b.business.telecomitalia.it,client,ipinhostname,clientwords]
-1.1 BAYES_05 BODY: Bayesian spam probability is 1 to 5%
[score: 0.0426]
0.0 UPPERCASE_25_50message body is 25-50% uppercase
0.2 AWLAWL: From: address is in the auto white-list
---BeginMessage---
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option d
(full-duplex), but I need to make ringing the phone in intercom.
Now I set auto-answer=6 but after first ring the phone hangup the call.
There is a way to using page/intercom with normal ring and not with
auto-answer?
My dialplan:
[ext-paging]
include = ext-paging-custom
exten = PAGE4441,1,GotoIf($[ ${CALLERID(number)} = 4441 ]?skipself)
exten = PAGE4441,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4441,n,Set(AVAILSTATUS=not checked)
exten = PAGE4441,n,Goto(SKIPCHECK)
exten = PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js)
exten = PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4441,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4441,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep))
exten = PAGE4441,n(skipself),Noop(Not paging originator)
exten = PAGE4441,n,Hangup
exten = PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = PAGE4442,1,GotoIf($[ ${CALLERID(number)} = 4442 ]?skipself)
exten = PAGE4442,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4442,n,Set(AVAILSTATUS=not checked)
exten = PAGE4442,n,Goto(SKIPCHECK)
exten = PAGE4442,n(AVAIL),ChanIsAvail(${DB(DEVICE/4442/dial)}|js)
exten = PAGE4442,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4442,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4442,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4442,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4442,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4442,n,Dial(${DB(DEVICE/4442/dial)},5, A(beep))
exten = PAGE4442,n(skipself),Noop(Not paging originator)
exten = PAGE4442,n,Hangup
exten = PAGE4442,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
exten = 4446,1,Set(_FORCE_PAGE=1)
exten = 4446,n,Macro(user-callerid,)
exten = 4446,n,Set(TIMEOUT(absolute)=60)
exten = 4446,n,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
---End Message---
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[asterisk-users] Conference Page question
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option d
(full-duplex), but I need to make ringing the phone in intercom.
Now I set auto-answer=6 but after first ring the phone hangup the call.
There is a way to using page/intercom with normal ring and not with
auto-answer?
My dialplan:
[ext-paging]
include = ext-paging-custom
exten = PAGE4441,1,GotoIf($[ ${CALLERID(number)} = 4441 ]?skipself)
exten = PAGE4441,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4441,n,Set(AVAILSTATUS=not checked)
exten = PAGE4441,n,Goto(SKIPCHECK)
exten = PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js)
exten = PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4441,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4441,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep))
exten = PAGE4441,n(skipself),Noop(Not paging originator)
exten = PAGE4441,n,Hangup
exten = PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = PAGE4442,1,GotoIf($[ ${CALLERID(number)} = 4442 ]?skipself)
exten = PAGE4442,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4442,n,Set(AVAILSTATUS=not checked)
exten = PAGE4442,n,Goto(SKIPCHECK)
exten = PAGE4442,n(AVAIL),ChanIsAvail(${DB(DEVICE/4442/dial)}|js)
exten = PAGE4442,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4442,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4442,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4442,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4442,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4442,n,Dial(${DB(DEVICE/4442/dial)},5, A(beep))
exten = PAGE4442,n(skipself),Noop(Not paging originator)
exten = PAGE4442,n,Hangup
exten = PAGE4442,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
exten = 4446,1,Set(_FORCE_PAGE=1)
exten = 4446,n,Macro(user-callerid,)
exten = 4446,n,Set(TIMEOUT(absolute)=60)
exten = 4446,n,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] *****SPAMZ***** Asterisk cluster - keep up connection?
Spam detection software, running on the system placebo, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see [EMAIL PROTECTED] for details. Content preview: Hi all, how can I set up an asterisk cluster (using SER or hearbeat) that keep up call and conference room when one of * server go in fail state? There is a way that I install my asterisk software on a virtual machine compose of N * server? [...] Content analysis details: (5.3 points, 4.0 required) pts rule name description -- -- 5.0 BOTNET Relay might be a spambot or virusbot [botnet0.7,ip=82.184.107.109,hostname=host109-107-static.184-82-b.business.telecomitalia.it,client,ipinhostname,clientwords] -0.7 BAYES_20 BODY: Bayesian spam probability is 5 to 20% [score: 0.0875] 1.0 SAGREY Adds 1.0 to spam from first-time senders ---BeginMessage--- Hi all, how can I set up an asterisk cluster (using SER or hearbeat) that keep up call and conference room when one of * server go in fail state? There is a way that I install my asterisk software on a virtual machine compose of N * server? -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PHP sip client
Hi all, I want to write a simit sip client in PHP with asterisk API, in this moment I'm able to compose a number on my browser and call between 2 hw sip phone. I digit a number, my phone ring and after hanging up the cornet the second phone ring. But I want to add a features I want to hang up the cornet of my phone, compose the number in my browser and call a second phone. In witch way I do this? Can i do this? quickly... I want to replicate the numeric keyboard of my hw phone! Thanks in advantage and sorry for my english. :( -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk HA
Hi all, I have to make for a client an asterisk system for process up to 250 calls between conference and normal call. At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client require a failover system. Anyone have experience for this type of solution? Is better ultramonkey, dundi or SER proxy in front of * server? Thanks Enrico P.S. Now during all this year I have to work with this type of solution, why not make a fork of this ml for example [EMAIL PROTECTED], for write some docs too. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] qualify=yes
hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes
Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. Eheheheh -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Nokia E60/61/70 and SIP
Martin Joseph wrote: For all of us using these devices, I have some good news. There is a self installable firmware update available from Nokia here (requires windows box to install): http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate This seems to radically improve the behavior of the SIP client on my E60. It seems to have resolved several of the MANY bugs that were outstanding on this product. The update does erase all your setups and info though. You are warned. Marty Hi, there are differences for the lang? I have the E60 in italian lang and the software update says that I have the last firmware (I don't think is the last firmware). I have a problem with this phone and Asterisk, my sip.conf is: [208] username=208 type=friend secret=1234 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes (I have try with neven too) notransfer=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Pasqu 208 disallow=all allow=alaw ## I have set my phone with this conf: ## Public user name: sip:[PhoneNumber]@:[youre asterisks servers ip] or sip:[EMAIL PROTECTED] Use Compression: No Registration: Always on or When Needed Use Security: No -- Proxy Server: sip:[youre asterisks servers ip] Realm: asterisk User Name: PhoneNumber Password: PIN Allow loose routing: Yes Transport Type: UDP Port: 5060 -- Registrar Server: sip:[youre asterisks servers ip] Realm: asterisk User Name: PhoneNumber Password: PIN Transport Type: UDP Port: 5060 #3 but the phone not work. In the log I read this: ### with nat = never -- SIP read from 151.38.43.46:19834: REGISTER sip:192.168.1.200 SIP/2.0 Route: sip:192.168.1.200;lr Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED];expires=3600 CSeq: 1126 REGISTER Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T Supported: sec-agree Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.99 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.99:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED] Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T CSeq: 1126 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.1.99:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED];tag=as29a9cc9f Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T CSeq: 1126 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2495e6cb Content-Length: 0 ### with nat = yes # -- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 151.38.43.46:20300: REGISTER sip:192.168.1.200 SIP/2.0 Route: sip:pasqu.zapto.org;lr Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED];expires=3600 CSeq: 1133 REGISTER Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU Supported: sec-agree Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.99 : 5060 (NAT) Transmitting (NAT) to 151.38.43.46:20300: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED] Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU CSeq: 1133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 151.38.43.46:20300: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED];tag=as6985bfa3 Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU CSeq: 1133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest algorithm=MD5,
[asterisk-users] Bad number - is not in inbound speed dial
Hi, what mean this voice message that asterisk say when I try to call an extension of another asterisk connected by IAX2 trunk? This problem exist only if I call from asterisk1 to asterisk2, vice versa all work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 trunk problem
Hi at all, I have make a IAX2 trunk over openvpn between [EMAIL PROTECTED] and trixbox. [EMAIL PROTECTED] have extension 200 to 299 and Trixbox 300 to 399 In my 2 box I set in outbound routing that if I call 7|XXX I want to use the IAX trunk. The call from Trixbox (ext 301) to [EMAIL PROTECTED] (2xx) work very well but the call from [EMAIL PROTECTED] (2xx) to Trixbox (3xx) not work my pbx say: Bad number. I think the configuration is the same in two box but the call in one direction not work and I don't know the problem. I link the iax debug: http://rafb.net/paste/results/od3x1l26.html and the sip debug: http://rafb.net/paste/results/sao6v569.html What I have to do now? Thanks a lot Pasqualotto Enrico email: pasqu AT linux.it || enrico AT pasqualotto.org web: http://www.pasqualotto.org skype: epasqualotto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
