Re: [asterisk-users] Snom vs Polycom
The problem 'I can place calls but no one can reach me' is our number one support question. Advising the user to check the DND As a general comment, the DND button on a decent phone should LIGHT UP when it's in use. On the Polycom 650, it is very clear on the LCD screen with flashing icons, but it would be much better to have the button lit when in use, and perhaps add a broken dial tone as well. On the opther hand, the button is not under the transfer button. /r -- _ -- 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] MYSQL grammar diff in 1.6.2.1?
hi, dear all
MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
is that the grammar is different between them?
extensions.conf
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
blockenabled = 1)
cli:
-- Executing [...@macro-checkblacklist:2] MYSQL(SIP/1003-0006,
Query resultid 1 SELECT\ callerid\ from\ blacklist\ where\
companycode = 95040654321 and callerid=1003 and blockenabled = 1) in
new stack
[Jan 25 17:05:34] WARNING[2583]: app_addon_sql_mysql.c:374
aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an
error in your SQL syntax; check the manual that corresponds to your
MySQL server version for the right syntax to use near '\ callerid\
from\ blacklist\ where\ companycode = 95040654321 and callerid=1003 '
at line 1
--
Best regards,
Sucan
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Re: [asterisk-users] MySQL RealTime Error
What happens when you try the command mysql -uroot -proot asterisk Ish Zhang Shukun wrote: hi,all when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql database anymore, error as follow: [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf) the content of res_mysql.conf is: http://www.pastebin.org/81966 i've try command mysql -uroot -proot ,i can connect to mysql successfully. Could you tell me what's wrong with me ? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Detected digit 'f'
Hi, Does anyone know what it means when I've got an incoming fax routed through to iaxmodem+hylafax and then I see this in the asterisk log: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Googling hasn't helped me here :( -- Cheers, Kingsley. -- _ -- 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] Web-Meetme 4.0 and Asterisk 1.6.2
Hi, I'm trying to setup Web-Meetme 4.0 and I always get the following warning when I open the default page http://localhost/web-meetme Warning: session_start() [function.session-start]: Cannot send session cache limiter - headers already sent (output started at /var/www/web-meetme/locale.php:36) in /var/www/web-meetme/meetme_control.php on line 34 Has anyone a solution to this? Cheers Joern -- _ -- 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
Re: [asterisk-users] Detected digit 'f'
Kingsley Tart wrote: Hi, Does anyone know what it means when I've got an incoming fax routed through to iaxmodem+hylafax and then I see this in the asterisk log: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This may be related: http://www.trixbox.org/forums/trixbox-forums/help/fax-detected-no-fax-extension-0 My Google search: asterisk Detected digit 'f' Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] Web-Meetme 4.0 and Asterisk 1.6.2
joern wrote: I'm trying to setup Web-Meetme 4.0 and I always get the following warning when I open the default page http://localhost/web-meetme Warning: session_start() [function.session-start]: Cannot send session cache limiter - headers already sent (output started at /var/www/web-meetme/locale.php:36) in /var/www/web-meetme/meetme_control.php on line 34 Solved... I've just deleted the last empty line in the locale.php file -- _ -- 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] queue
Hi, all Is ther any way to pass channel queue such a way Queue(SIP/1001SIP/1002SIP/1003) thanks, Bhrugu Mehta -- _ -- 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
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- 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 -- _ -- 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] sip.conf with versatel and two NICs very strange problem
Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- 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
Re: [asterisk-users] Using SIPPEER status with CUT function?
JR Richardson wrote:
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
That is a bug; the function should be returning OK without the
calculated lag value.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
thanks, i tried this already but unfortunately no change. any further suggestions or answers concerning my other questions? thanx, yves Cary Fitch schrieb: As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- 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
Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem
- Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y Either a typo or you have an IP conflict? --Tim -- _ -- 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] Call recordings and sensitive information
During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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] Call tagging
Something similar along the lines of a previous email - has anyone developed, or is using, something similar to this http://www.veritape.com/wp-content/uploads/2009/11/veritape-call-tagging-module-description.pdf Julian -- _ -- 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
Re: [asterisk-users] Call recordings and sensitive information
Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can pause and un-pause monitoring. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Monday, January 25, 2010 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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
Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem
thanx... a typo... the routers local ip is 10.26.208.253 yves Tim Nelson schrieb: - Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y Either a typo or you have an IP conflict? --Tim -- _ -- 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
Re: [asterisk-users] ivvr with asterisk
Yes, you can using SIP *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: qu...@vega.com.vn To: asterisk-users@lists.digium.com Date: Mon, 25 Jan 2010 08:35:31 +0700 Subject: Re: [asterisk-users] ivvr with asterisk Thanks all, Before purchasing any device i want to make some prototype of IVVR, is it possible to use asterisk to build an IVVR with softphones (such as SIP softphone)? and Is there any example about these? Quyps On Sat, 2010-01-23 at 11:44 +0530, mtha...@gmail.com wrote: Quyps, It looks like you mis-read the picture. Asterisk is the core, it need to be there regardless you use FreePBX or Tribox. FreePBX is a GUI web interface to manage asterisk. Itself is not an IP-PBX. Trixobx, still based on the Asterisk + freePBX, adds some more additional applications based on the community feed back and requirement. Trixbox is an easy go, but there may be some unwanted stuff with it. elastix.org is also a nice package, give it a try. Regards MT Kondela kevesystems.com On Sat, Jan 23, 2010 at 7:32 AM, Pham Quy qu...@vega.com.vn wrote: Hi all, First I'm very new. I want to build an Interactive Video-voice Response system. There is number of choice I have found so far: FreePBX, TriBox, Asterisk. Which is the best in my case? and what do i need to build such IVVR system? Thanks. Quyps -- _ -- 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 -- _ -- 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 -- _ -- 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 _ -- _ -- 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
Re: [asterisk-users] queue
From what I read, queue is agent-specific, not channel (I've only been playing with this for two days, so don't jump too hard, gurus.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta Sent: Monday, January 25, 2010 6:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] queue Hi, all Is ther any way to pass channel queue such a way Queue(SIP/1001SIP/1002SIP/1003) thanks, Bhrugu Mehta -- _ -- 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
Re: [asterisk-users] fax over IP - http/ftp-provisioning - intercom
On Sat, 2010-01-23 at 21:19 -0500, Alex Balashov wrote: What is the situation with Asterisk and fax over IP ? Can Asterisk receive a fax over a POTS or ISDN line ?? Do I then need a Digium TDM-card and an FXO-module or a T38-gateway ? Despite what anyone may say about Fax over IP allegedly works for them, save yourself the trouble and make sure you take the POTS and ISDN approach. If I keep the POTS-line or the ISDN-line, can Asterisk then transform an incoming fax to an email with pdf or tiff attachment ?? And the other way around, can an email to the Asterisk-server be transformed to an analogue fax ?? Jonas. -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Here are my version levels: Asterisk 1.6.2.1 built by root on a x86_64 running Linux on 2010-01-15 16:22:39 UTC Linux 2.6.18-164.11.1.el5 #1 SMP Wed Jan 6 13:26:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux MARK. -- _ -- 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] Problem with Digium card, not transfering outgoing calls
Hi, I'm experiencing some strange problems with out Digium card. First the details abount hardware and software: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) Asterisk 1.6.0.20 dahdi-linux-complete-2.2.1-rc2+2.2.1-rc2 libpri-1.4.10.2.tar.gz The problem now is that there are a number of clients that I call and suddenly the connection drops. Just a few seconds later the clients calls me, and tells me that he has accepted the calls but didn't hear anything. The CLI output for one of the calls is: Using SIP RTP CoS mark 5 -- Executing [01777622...@local:1] Set(SIP/sguenther-0016, CALLERID(num)=8304498) in new stack -- Executing [01777622...@local:2] Dial(SIP/sguenther-0016, DAHDI/g1/01777622XXX,60,tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/01777622700 -- DAHDI/10-1 is proceeding passing it to SIP/sguenther-0016 -- DAHDI/10-1 is making progress passing it to SIP/sguenther-0016 -- DAHDI/10-1 is making progress passing it to SIP/sguenther-0016 -- DAHDI/10-1 is ringing [2010-01-25 09:11:11] ERROR[20610]: chan_dahdi.c:10594 dahdi_pri_error: XXX Message longer than it should be?? XXX -- Channel 0/1, span 4 got hangup request, cause 16 -- Channel 0/1, span 4 received AOC-E charging 5 units -- Hungup 'DAHDI/10-1' -- No one is available to answer at this time (1:0/0/0) -- Executing [01777622XXXlocal:3] Hangup(SIP/sguenther-0016, ) in new stack I have found a number of postings about XXX Message longer than it should be?? XXX but I guess these problems have been fixed in the current versions. Again: It is no problem to accept calls and the majority of outgoing calls are no problem. Thanks for any suggestions or hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- _ -- 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
Re: [asterisk-users] Detected digit 'f'
Kingsley Tart wrote: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Turn off fax detection. Thanks, Lee. -- _ -- 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
Re: [asterisk-users] queue
On Mon, Jan 25, 2010 at 6:10 AM, bhrugu mehta mehtabhr...@gmail.com wrote: Hi, all Is ther any way to pass channel queue such a way Queue(SIP/1001SIP/1002SIP/1003) thanks, Bhrugu Mehta You would define those SIP peers as members in queues.conf: [queue_name] member = SIP/1001 member = SIP/1002 member = SIP/1003 and you would call the queue from the dialplan using the following: exten = 1000,1,Queue(queue_name) Read through the sample queues.conf for a listing of all the queue-specific options you can set, and also do a core show application queue in the CLI to see which options are available for the Queue() command in the dialplan. -- Thanks, --Warren Selby http://www.selbytech.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
Re: [asterisk-users] Call recordings and sensitive information
Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Call recordings and sensitive information
Oh, crap. the second I send, I realize I use features.conf, right ? ;) Is there any other way of getting this into the dialplan ? I would rather not have to have the users pressing a key, but for software to intercept the appropriate point and perform some AMI command Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Call recordings and sensitive information
This is crazy. Something about writing to the list gives me ideas ;) What I am looking for is show manager command pausemonitor ;) Thanks anyway, all. Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Oh, crap. the second I send, I realize I use features.conf, right ? ;) Is there any other way of getting this into the dialplan ? I would rather not have to have the users pressing a key, but for software to intercept the appropriate point and perform some AMI command Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Detected digit 'f'
On Mon, 2010-01-25 at 07:50 -0800, Lee Howard wrote: Kingsley Tart wrote: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Turn off fax detection. Hi Lee, Thanks for your reply. Fax detection is already turned off in chan_dahdi.conf :( I had initially just commented out the lines that had turned it on but I have tried adding specific faxdetect=no lines but that has made no difference. The weird thing is that I can receive faxes from a friend's fax server but not from the fax machine here in the office (whereupon I have the above problem). -- Cheers, Kingsley. -- _ -- 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
Re: [asterisk-users] Call recordings and sensitive information
Glad we could (not) help. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Monday, January 25, 2010 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call recordings and sensitive information This is crazy. Something about writing to the list gives me ideas ;) What I am looking for is show manager command pausemonitor ;) Thanks anyway, all. Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Oh, crap. the second I send, I realize I use features.conf, right ? ;) Is there any other way of getting this into the dialplan ? I would rather not have to have the users pressing a key, but for software to intercept the appropriate point and perform some AMI command Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can pause and un-pause monitoring. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Monday, January 25, 2010 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Detected digit 'f'
On Mon, 2010-01-25 at 05:49 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Hi,
Does anyone know what it means when I've got an incoming fax routed
through to iaxmodem+hylafax and then I see this in the asterisk log:
DEBUG[18902] chan_dahdi.c: Detected digit 'f'
This may be related:
http://www.trixbox.org/forums/trixbox-forums/help/fax-detected-no-fax-extension-0
My Google search:
asterisk Detected digit 'f'
Hi,
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going on.
In our application it's the dialled number that decides whether to jump
to the recvfax priority, hence all numeric extensions starting with zero
being allowed.
This is what I've got in extensions.conf for receiving faxes (as a test
I added a few specific extensions f and fax to the end of the
recvfax context but those lines never get executed):
[recvfax]
exten = _0.,1,SET(__channel2=${UNIQUEID})
exten = _0.,2,AGI(service_nts_nextgen|answered|${channel1}|${channel2}|7)
exten = _0.,3,Set(INIT_EXTEN=${EXTEN})
exten =
_0.,4,Set(__FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)}-${INIT_EXTEN})
exten = _0.,5,SET(CALLERID(name)=${FAXFILE})
exten = _0.,6,Dial(IAX2/iaxmodem0/${INIT_EXTEN})
exten = _0.,7,Dial(IAX2/iaxmodem1/${INIT_EXTEN})
exten = _0.,8,Dial(IAX2/iaxmodem2/${INIT_EXTEN})
exten = _0.,106,NoOp(test2)
exten = h,1,System('/var/lib/asterisk/agi-bin/fax2email.pl --source-file
${FAXFILE}.tif --format ${fax2email_format} --paper-size
${fax2email_paper_size} --email-to ${fax2email_to_addr} --email-from
${fax2email_from_addr} --email-from-name $
{fax2email_from_name} --cli ${fax2email_cli} --dni ${fax2email_dni}')
exten =
h,2,deadAGI(service_nts_nextgen|completed|${channel1}|ANSWER|${UNIQUEID}|0)
exten = fax,1,NoOp(fax extension)
exten = fax,2,Hangup
exten = f,1,NoOp(f extension)
exten = f,2,Hangup
This is what I get in the asterisk log (I've just extracted what I
thought was the relevant bit to keep this email short):
[Jan 25 16:41:58] VERBOSE[3567] logger.c: -- Executing
[08454632...@recvfax:6] Dial(DAHDI/59-1, IAX2/iaxmodem0/08454632501) in new
stack
[Jan 25 16:41:58] DEBUG[3567] chan_iax2.c: prepending 8 to prefs
[Jan 25 16:41:58] VERBOSE[3567] logger.c: -- Called iaxmodem0/08454632501
[Jan 25 16:41:58] VERBOSE[3470] logger.c: -- Call accepted by 127.0.0.1
(format alaw)
[Jan 25 16:41:58] VERBOSE[3470] logger.c: -- Format for call is alaw
[Jan 25 16:41:58] VERBOSE[3567] logger.c: -- IAX2/iaxmodem0-4643 is ringing
[Jan 25 16:41:58] VERBOSE[3567] logger.c: -- IAX2/iaxmodem0-4643 answered
DAHDI/59-1
[Jan 25 16:41:58] DEBUG[3567] chan_dahdi.c: Detected digit 'f'
[Jan 25 16:42:31] VERBOSE[3493] logger.c: -- Channel 0/28, span 2 got
hangup request, cause 16
When my friend sends a fax that works, I don't get the Detected digit
'f' line in the log. Not sure whether that suggest anything?
--
Cheers,
Kingsley.
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Re: [asterisk-users] MySQL RealTime Error
You must read the upgrade instructions. The database definitions in res_mysql.conf have changed. The way you reference the database in extconfig.conf is also different. On Mon, 2010-01-25 at 09:33 +, Ishfaq Malik wrote: What happens when you try the command mysql -uroot -proot asterisk Ish Zhang Shukun wrote: hi,all when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql database anymore, error as follow: [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf) the content of res_mysql.conf is: http://www.pastebin.org/81966 i've try command mysql -uroot -proot ,i can connect to mysql successfully. Could you tell me what's wrong with me ? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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
Re: [asterisk-users] Snom vs Polycom
From: cb c...@mythtech.net Sent: Sunday, January 24, 2010 12:42 I use the Snom 370 all day long at work. I have never had a problem adjusting the volume. I change it multiple times a day as I keep my handset on one volume and my headset on another, so I'm always going up and down and I've never accidentally pressed any other key. I will however agree with you on the Mute button, any time I want to mute a call, I have to stop and look at the buttons and figure out which one of the tiny ones is mute. You are right, especially with practice. I still think it's a little easier if the buttons are distinct from one another by appearance, size and/or placement.. I think Polycom made a good design decision by not making you 'reach over' any buttons to press these common buttons. I also like the fact that the mute button turns bright red when activated. Come to think of it, I wish the DND button turned red when activated :-). Again, the Snom allows pretty liberal programming so it would likely be possibly to disable that button from doing DND and change to something else less likely to accidentally be hit. However that won't resolve the issue of the display icon being so small allowing you to easily forget you turned it on in the first place. Everyone bitches about Polycom for not having a full-featured web gui for provisioning. These 'bitchings' have significant merit in the case of 'trying out' Polycom phones, or for installations with a small number of Polycom phones. Having said that, the Polycom provisioning system is simple and powerful. IMO it makes things very easy in the long run even if you just 2 or 3 phones . It allows you to 'layer' your configs so you end up with settings for everyone, settings for groups, and settings for phone x,y, and z. When you need to change any an aspect of programming everywhere, for a group, or for just one phone, it's a trivial and non-iterative matter to do so. D.R.Y. Firmware updates are as simple as dropping the new release onto your ftp/tftp/http provisioning server, but progressive releases are also possible. It's well thought out. I'm guessing you are talking about using the X button here. Weird that yours plays the dial tone again right away as mine does not. I wonder if this could be a difference in firmware versions. I believe I am currently running 7.3.24 although I could be wrong (but it should be one around there). I had a conversation with Mike Storella and one of Snom's developers describing why this characteristic is clumsy for several use cases. Even though it didn't make sense for us to use their phones at that point, I really had to tip my hat to Snom because they really were engaged in understanding our viewpoint 'from the trenches'. So yes Snom is very receptive and hopefully they did fix it in firmware, or at least between the 3xx and 8xx series phones. One thing that does drive me nuts about my Snom, if I am on a call, and have another ringing, I can't change to a different talk method (handset/headset/speaker) until the ringing call on call waiting stops. If I try, it will automatically put the first call on hold and answer the 2nd call. I've had this happen several times as I change from handset to headset mid call and find myself suddenly talking to a different caller. Hear! Hear! So far every phone I've ever used has some significant idiosyncrasies (idioiticsyncracies), but I guess that's the purpose of these subjective shakedowns--to determine the cost-benefit analysis of various options. I'm very hopeful about the Aastra 6739i. http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-82EE0F6D/04/6739i_pds_en_1209.pdf. Supposedly available within a few weeks. -Karl -- _ -- 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] [OT] spa3000 (Regional Line1) NL settings required
Dear all, Can someone from Netherlands who has SPA3000 send me the Regional Line1 settings. I'm not sure what is wrong but I can call from asterisk to phone attached to spa300 FXO, but not the other way. I tested three phones: siemens gigaset, tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk, other two just dial but nothing happens. I guess this has something with the Regional settings?? Best regards, Lukasz -- _ -- 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] How to make SpeechBackground keep playing if utterance doesn't match our grammar
Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio—basically, like a podcast—and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition engine.) Problem is, any speech is always matching something, even if it matches with a low score. This is going to be annoying to our users, as you can imagine. A second, only partly related problem is that we'd like to have the fast-forward and rewind abilities of ControlPlayback, but with speech recognition. We have a hackish way around this—make a sound file starting several seconds into the first file, and call SpeechBackground on that— but it's a kludge. So what we really want is something that acts like SpeechBackground, but blocks until there's a matching result (preferrably with rewind and fast-forward support). The first part, block if no match, is essential; the second part, rewind and fast-forward, is not. I know this is a tall order, but, hey, it can't hurt to ask, right? Has anyone faced this before? Is there any application or AGI command that works? (We couldn't find one.) Are we going to have to bust out gcc and write our own dialplan application? :/ Thanks a lot, -- Quinn Weaver Consulting, LLC Full-stack web design and development http://quinnweaver.com/ 510-520-5217 -- _ -- 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
Re: [asterisk-users] How to make SpeechBackground keep playing ifutterance doesn't match our grammar
I assume you've tried the $GARBAGE in your grammar to make speech run tightly? -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver Sent: Monday, January 25, 2010 2:58 PM To: asterisk-speech-rec-requ...@lists.digium.com Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] How to make SpeechBackground keep playing ifutterance doesn't match our grammar Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio-basically, like a podcast-and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition engine.) Problem is, any speech is always matching something, even if it matches with a low score. This is going to be annoying to our users, as you can imagine. A second, only partly related problem is that we'd like to have the fast-forward and rewind abilities of ControlPlayback, but with speech recognition. We have a hackish way around this-make a sound file starting several seconds into the first file, and call SpeechBackground on that- but it's a kludge. So what we really want is something that acts like SpeechBackground, but blocks until there's a matching result (preferrably with rewind and fast-forward support). The first part, block if no match, is essential; the second part, rewind and fast-forward, is not. I know this is a tall order, but, hey, it can't hurt to ask, right? Has anyone faced this before? Is there any application or AGI command that works? (We couldn't find one.) Are we going to have to bust out gcc and write our own dialplan application? :/ Thanks a lot, -- Quinn Weaver Consulting, LLC Full-stack web design and development http://quinnweaver.com/ 510-520-5217 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] How to make SpeechBackground keep playing ifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 1:06 PM, Danny Nicholas da...@debsinc.com wrote:
I assume you've tried the $GARBAGE in your grammar to make speech run
tightly?
Yes, we use $GARBAGE currently. Here's our grammar:
#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format lumenvox/1.0;
root $Command;
$Play_Next = [play] next $GARBAGE;
$Quit = quit $GARBAGE;
$Rewind = rewind $GARBAGE;
$Previous = (previous | privius | preevious) $GARBAGE;
$Pause = pause $GARBAGE;
$Fast_Forward = ([fast] forward) $GARBAGE;
$Replay = replay $GARBAGE;
$Command =
$Play_Next {$ = play_next;}
| $Quit {$ = quit;}
| $Rewind {$ = rewind;}
| $Previous {$ = previous;}
| $Pause {$ = pause;}
| $Fast_Forward {$ = fast_forward;}
| $Rewind {$ = rewind;}
| $Replay {$ = replay;};
Thanks,
[Omitting asterisk-speech-rec due to some problems with Mailman]
--
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Full-stack web design and development
http://quinnweaver.com/
510-520-5217
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Re: [asterisk-users] How to make SpeechBackground keep playingifutterance doesn't match our grammar
What does your dialplan snippet to run this look like?
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver
Sent: Monday, January 25, 2010 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SpeechBackground keep
playingifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 1:06 PM, Danny Nicholas da...@debsinc.com wrote:
I assume you've tried the $GARBAGE in your grammar to make speech run
tightly?
Yes, we use $GARBAGE currently. Here's our grammar:
#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format lumenvox/1.0;
root $Command;
$Play_Next = [play] next $GARBAGE;
$Quit = quit $GARBAGE;
$Rewind = rewind $GARBAGE;
$Previous = (previous | privius | preevious) $GARBAGE;
$Pause = pause $GARBAGE;
$Fast_Forward = ([fast] forward) $GARBAGE;
$Replay = replay $GARBAGE;
$Command =
$Play_Next {$ = play_next;}
| $Quit {$ = quit;}
| $Rewind {$ = rewind;}
| $Previous {$ = previous;}
| $Pause {$ = pause;}
| $Fast_Forward {$ = fast_forward;}
| $Rewind {$ = rewind;}
| $Replay {$ = replay;};
Thanks,
[Omitting asterisk-speech-rec due to some problems with Mailman]
--
Quinn Weaver Consulting, LLC
Full-stack web design and development
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510-520-5217
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Re: [asterisk-users] How to make SpeechBackground keep playingifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 2:07 PM, Danny Nicholas da...@debsinc.com wrote:
What does your dialplan snippet to run this look like?
It's part of a Perl FastAGI, running on a separate box from Asterisk.
The Perl code is using Asterisk::AGI's exec() method to call
SpeechBackground:
my $sb_retval = $c-agi-exec('SpeechBackground', $path);
where $path specifies a .sln file on the remote (Asterisk) host.
Here's the dialplan snippet for invoking the FastAGI. Some notes:
- 127.0.0.1:4575 is an ssh tunnel to my box, where I'm doing the
development. I know this works; I hear sound, DTMF works, speech
recognition occurs, et cetera.
- While my dialplan does Answer() and Hangup(), my Perl program does
it as well. Doesn't seem to cause a problem.
- The extension name, XX was my client's actual phone number,
so I X'ed it out.
[inbound]
exten = XX,s,Answer()
exten = XX,1,Background(demo-congrats)
exten = XX,n,Hangup()
...
exten = 3,1,Goto(quinn-tunnel-test,1,1)
exten = 3,n,Hangup()
[quinn-tunnel-test]
exten = 1,1,Answer()
exten = 1,n,AGI(agi://127.0.0.1:4575/Entry/entry?)
exten = 1,n,Hangup()
--
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Re: [asterisk-users] How to make SpeechBackground keepplayingifutterance doesn't match our grammar
Since you're Perling it, why not just put the $sb_retval in a while loop
like this:
- my $response_good=0;
- my $sb_retval=undef;
- while (! $response_good) {
-my $tmp_retval = $c-agi-exec('SpeechBackground', $path);
-if ($tmp_retval eq 'play_next') {
$sb_retval=$tmp_retval;
$response_good=1;
}
...
}
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver
Sent: Monday, January 25, 2010 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SpeechBackground
keepplayingifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 2:07 PM, Danny Nicholas da...@debsinc.com wrote:
What does your dialplan snippet to run this look like?
It's part of a Perl FastAGI, running on a separate box from Asterisk.
The Perl code is using Asterisk::AGI's exec() method to call
SpeechBackground:
my $sb_retval = $c-agi-exec('SpeechBackground', $path);
where $path specifies a .sln file on the remote (Asterisk) host.
Here's the dialplan snippet for invoking the FastAGI. Some notes:
- 127.0.0.1:4575 is an ssh tunnel to my box, where I'm doing the
development. I know this works; I hear sound, DTMF works, speech
recognition occurs, et cetera.
- While my dialplan does Answer() and Hangup(), my Perl program does
it as well. Doesn't seem to cause a problem.
- The extension name, XX was my client's actual phone number,
so I X'ed it out.
[inbound]
exten = XX,s,Answer()
exten = XX,1,Background(demo-congrats)
exten = XX,n,Hangup()
...
exten = 3,1,Goto(quinn-tunnel-test,1,1)
exten = 3,n,Hangup()
[quinn-tunnel-test]
exten = 1,1,Answer()
exten = 1,n,AGI(agi://127.0.0.1:4575/Entry/entry?)
exten = 1,n,Hangup()
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Full-stack web design and development
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510-520-5217
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[asterisk-users] Disa not fully bridging outbound call
Hello, I have a situation where a remote worker dials in to the asterisk server, enters the secret code, then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] -- Moving call from channel 21 to channel 2 [Jan 25 17:51:40] -- Zap/0:2-1 answered Zap/1-1 [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] -- Native bridging Zap/1-1 and Zap/0:2-1 [Jan 25 17:51:49] -- Channel 0/1, span 1 got hangup request, cause 16 [Jan 25 17:51:49] -- Hungup 'Zap/0:2-1' [Jan 25 17:51:49] == Spawn extension (from-inside-redir, 16037649936, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Executing [...@from-inside-redir:1] Hangup(Zap/1-1, ) in new stack [Jan 25 17:51:49] == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Hungup 'Zap/1-1' [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found? [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/2 on span 1 This says it is using DAHDI but it is actually still Zaptel as I have not had much success getting DAHDI to work on OpenSuSE, but that is another post for a later date. Any help is greatly appreciated. Thank You -- JohnM -- _ -- 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
Re: [asterisk-users] How to make SpeechBackground keepplayingifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 2:56 PM, Danny Nicholas da...@debsinc.com wrote:
Since you're Perling it, why not just put the $sb_retval in a while loop
like this:
- my $response_good=0;
- my $sb_retval=undef;
- while (! $response_good) {
- my $tmp_retval = $c-agi-exec('SpeechBackground', $path);
- if ($tmp_retval eq 'play_next') {
$sb_retval=$tmp_retval;
$response_good=1;
}
...
}
If we did that, we'd be replaying $path from the beginning every time
the user said something that didn't match the grammar. For a podcast
episode like a radio show, that's bad—you don't want to be 30 seconds
or two minutes into the content and have to start over.
Also, as I said, it's always matching one of the rules in our
grammar--even if I literally say goobledegook. So it's unclear how
we'd implement $response_good.
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http://quinnweaver.com/
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Re: [asterisk-users] MYSQL grammar diff in 1.6.2.1?
On Monday 25 January 2010 03:12:08 Zhang Shukun wrote:
hi, dear all
MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
is that the grammar is different between them?
extensions.conf
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
blockenabled = 1)
cli:
-- Executing [...@macro-checkblacklist:2] MYSQL(SIP/1003-0006,
Query resultid 1 SELECT\ callerid\ from\ blacklist\ where\
companycode = 95040654321 and callerid=1003 and blockenabled = 1) in
new stack
[Jan 25 17:05:34] WARNING[2583]: app_addon_sql_mysql.c:374
aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an
error in your SQL syntax; check the manual that corresponds to your
MySQL server version for the right syntax to use near '\ callerid\
from\ blacklist\ where\ companycode = 95040654321 and callerid=1003 '
at line 1
I have no idea why you backslashed your spaces in 1.4, as that has never been
a requirement (as is evident later in the line, where you neglected them).
This is the problem, and if you remove the backslashes, you should be fine.
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Re: [asterisk-users] Using SIPPEER status with CUT function?
On Monday 25 January 2010 07:43:30 Kevin P. Fleming wrote:
JR Richardson wrote:
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
That is a bug; the function should be returning OK without the
calculated lag value.
I don't believe that's true. At any rate, it's been this way for a very long
time, including 1.2, and it's probably inadvisable to change it now (as people
are probably depending upon its current behavior).
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Re: [asterisk-users] ASTSBINDIR not being picked up by safe_asterisk
On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. -- Tilghman -- _ -- 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] StopPlayTones() after first digit?
I configured our SIP gateway to automatically dial extension s when a phone is picked up. I want Asterisk to play a dial tone, wait for an extension to be dialled, and hangup on timeout This works great, but I also want Asterisk to *stop* playing the dial tone after the first digit is pressed So far my extensions.conf contains, [internal] exten = s,1,Answer exten = s,n,PlayTones(dial) exten = s,n,WaitExten How can I continue playing a dial tone as long as a digit isn't pressed (and TIMEOUT() hasn't expired) - but stop playing the dial tone as soon as the first digit is pressed? I think the Background() application works like this - it stops playing as soon as the first digit is pressed - it seems PlayTones() works differently? -- _ -- 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 1.2.37 + BLF + ParkedCalls + SPA962
Greetings all. First off, thank you for your time on this. I have spent literally 12 hours searching every forum and article I can find, and I'm going cross-eyed, so I need to bother everyone with this. I am running * 1.2.37, and I am trying to get the hints working, so I can turn one of my SPA962's LED's red when someone parks a call. I have used Button #3 on my SPA962 to successfully monitor Zap channels, SIP channels, trunks.. basically, everything ELSE, so I am pretty sure my phone is set up properly. But I CANNOT get this puppy to show me when someone parks a call, regardless of the fact that call parking also works perfectly on this system. I have found so much conflicting information regarding the [parkedcalls] context, and the hints entry for extensions.conf (to use exten=701,hint,park:7...@parkedcalls vs. exten=701,hint,Local/7...@parkedcalls), as well as how to set up features.conf, I just don't know which end is up. Does anyone out there know how to make this work with my setup? When I type show hints, earlier is was saying State:Unavailable, now it shows State:Idle but never does anything when I park a call. Here's the files: Features.conf: [general] parkext = 700 parkpos = 701-710 context = parkedcalls parkingtime = 90 Extensions.conf (there is more after this, but it's not relevant): [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] [default] include=parkedcalls exten = 701,1,ParkedCall(701) exten = 701,hint,Local/7...@parkedcalls CLI show hints: -= Registered Asterisk Dial Plan Hints =- 701 : Local/7...@parkedcall State:Idle Watchers 1 So, there you have it. I hope somebody can help. THANK YOU!! ~Joel -- _ -- 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
Re: [asterisk-users] How to make SpeechBackground keepplayingifutterance doesn't match our grammar
Quinn--
I would venture to guess that your problem is because you are using the
sound file streaming
mechanism at too high a level. At the app/agi level, you don't get any
control over the process.
You start the sound process, then you wait for the interrupt; it's all
neatly bundled into a single
package.
At the C level, using the machinery that Asterisk uses, you use these calls:
res = ast_streamfile(chan, Filename, chan-language);
if (res2) { failure_code(); }
else {
ast_autoservice_start(chan); /* keep those packets running out to chan
*/
}
/* put code here to process the stuff from voice recognition s/w until you
get what
you want, or time out, or whatever */
if (!res2) {
ast_stopstream(chan);
ast_autoservice_stop(chan);
}
The code above pretty much supposes that the file will play longer than it
will take to get some outcome
from the VR stuff; but no matter, if it runs clear to the end, it will just
leave you with a dead channel.
Best to set some sort of timeout so as not to sit forever if the person at
the other end of chan is mute
or dies or something. The main thing to remem is that autoservice_start()
and streamfile() immediately
return, and do not block, and the shuffling of sound packets is handled in a
different thread. Some other
event or timeout or something needs to eat up time between the starting of
the playback and the stopping
of the playback.
Hope this helps, probably not what you were hoping for.
murf
On Mon, Jan 25, 2010 at 4:33 PM, Quinn Weaver qu...@fairpath.com wrote:
On Mon, Jan 25, 2010 at 2:56 PM, Danny Nicholas da...@debsinc.com wrote:
Since you're Perling it, why not just put the $sb_retval in a while
loop
like this:
- my $response_good=0;
- my $sb_retval=undef;
- while (! $response_good) {
-my $tmp_retval = $c-agi-exec('SpeechBackground', $path);
-if ($tmp_retval eq 'play_next') {
$sb_retval=$tmp_retval;
$response_good=1;
}
...
}
If we did that, we'd be replaying $path from the beginning every time
the user said something that didn't match the grammar. For a podcast
episode like a radio show, that's bad—you don't want to be 30 seconds
or two minutes into the content and have to start over.
Also, as I said, it's always matching one of the rules in our
grammar--even if I literally say goobledegook. So it's unclear how
we'd implement $response_good.
--
Quinn Weaver Consulting, LLC
Full-stack web design and development
http://quinnweaver.com/
510-520-5217
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Steve Murphy
ParseTree Corp
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