[asterisk-users] Asterisk Queue Dialplan
Hi, I want to reconfigure my asterisk dialplan.I have a problem.I have 4 agents in a queue.How is the configuration for the asterisk dialplan if I want to have only 4 agents maximum who can receive the phone,so if the fifth caller try to entering the queue they will be noted by my IVR that all our agents are busy?Thank you so much for this millis,it really helpful especially for a newbie like me. Best Regards, Daniel ___ -- 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] DEVICE_STATE - Solved
Thx, that did the trick!
On Sat, 12 Dec 2009 17:34:19 +0100, Philipp Kempgen wrote:
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,NoOp(0317998...@inputinterior.se has state
${DEVICE_STATE(SIP/0317998975)})
exten = 0317998975,2,Dial(SIP/0317998975)
It doesn't matter if I have a call on 0317998975 or not. i always get:
-- Executing [0317998...@inputinterior.se:1]
NoOp(SIP/0317998985-0011, 0317998...@inputinterior.se has state
NOT_INUSE) in new stack
So I figure out that I have missed something but cant figure out what.
:(
Any ideeas?
sip.conf:
[general]
allowsubscribe = yes
notifyringing = yes
notifyhold = yes
limitonpeers = yes
Philipp Kempgen
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[asterisk-users] Avaya 9650 SIP phone and dial timeout
Hi!
Have a weired problem with Avaya 9650 phones:
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 0317998975,2,Hangup()
exten = 0317998975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten = 0317998975-INUSE,2,Hangup()
exten = 0317998975-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,us)
exten = 0317998975-NOANSWER,2,Hangup()
exten = 0317998975-NOT_INUSE,1,Dial(SIP/0317998975,2)
exten = 0317998975-NOT_INUSE,2,Goto(0317998975-${DIALSTATUS},1)
exten = 0317998975-NOT_INUSE,3,Hangup()
I know that I have a very short dial timeout, just for testing purposes.
If i call 0317998975 and that extension is free:
The 9650 phones rings for 2 seconds.
== Using UDPTL CoS mark 5
-- Executing [0317998...@inputinterior.se:1]
Goto(SIP/0317998985-0031, 0317998975-NOT_INUSE,1) in new stack
-- Goto (inputinterior.se,0317998975-NOT_INUSE,1)
-- Executing
[0317998975-not_in...@inputinterior.se:1]
Dial(SIP/0317998985-0031, SIP/0317998975,2) in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- SIP/0317998975-0032 is ringing
-- Nobody picked up in 2000 ms
-- Executing [0317998975-not_in...@inputinterior.se:2]
Goto(SIP/0317998985-0031, 0317998975-NOANSWER,1) in new stack
-- Goto (inputinterior.se,0317998975-NOANSWER,1)
-- Executing [0317998975-noans...@inputinterior.se:1]
VoiceMail(SIP/0317998985-0031, 0317998...@inputinterior.se,us) in
new stack
-- Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
And as u can see the systems plays my unavailable message but,
the 9650 phones keep ringing, forever, or at least until I lift
and put down the handset.
Any ideas how i cant stop the ringing?
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[asterisk-users] iphone client app
Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex signature.asc Description: Digital 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
Re: [asterisk-users] iphone client app
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I have not done any Asterisk-specific testing, I hope someone who has will chime in. /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
Re: [asterisk-users] iphone client app
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 13/12/2009 10:24, Randy R wrote: On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I have not done any Asterisk-specific testing, I hope someone who has will chime in. iSip (£2.39) http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8 SIP Softphone (£4.99) http://itunes.apple.com/gb/app/acrobits-softphone-sip-phone-for/id314192799?mt=8 WeePhone SIP (£2,99) http://itunes.apple.com/gb/app/weephone-sip/id301500729?mt=8 Each of the above are SIP based, I have seen 1 or 2 IAX soft-phones in the AppStore, but never used them. I have used both iSip SIP Soft-phone both with good results on my iPod Touch. iSip does have one advantage that is does support 'Push' notifications. I have not used WeePhone SIP, just found that one will looking for the URL of the other 2. Hope that helps. - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkskxgIACgkQvp6arS3vDiptSQCfZog+f9ieFlxcwOHpjFtCY8kw UmUAoKj58yrbcrxwBggWSKKeAqYotNpy =oBao -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- 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] Unable to open file...
Removing the spaces did it. I works now. I used the space for clarity but, Asterisk didn't like it. Thanks for your time. --- On Sat, 12/12/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] Unable to open file... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, December 12, 2009, 10:38 PM Take the whitespace out of your ()'s. It's: exten = 80,n,BackGround(es/good) not exten = 80,n,BackGround( es/good ) Thanks, --Warren Selby On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com wrote: Same thing: == Using SIP RTP CoS mark 5 -- Executing [...@outbound:1] Answer(SIP/102-096a48c8, ) in new stack -- Executing [...@outbound:2] Verbose(SIP/102-096a48c8, In timeofday ) in new stack In timeofday -- Executing [...@outbound:3] GotoIfTime(SIP/102-096a48c8, 00:00-12:00,*,*,*?day) in new stack -- Executing [...@outbound:4] GotoIfTime(SIP/102-096a48c8, 12:01-17:59,*,*,*?afternoon) in new stack -- Executing [...@outbound:5] GotoIfTime(SIP/102-096a48c8, 18:00-11:59,*,*,*?night) in new stack -- Goto (outbound,80,16) -- Executing [...@outbound:16] Verbose(SIP/102-096a48c8, Night..) in new stack Night.. -- Executing [...@outbound:17] BackGround(SIP/102-096a48c8, es/ good ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/good does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/good (format 0x8 (alaw)): No such f ile or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/good -- Executing [...@outbound:18] BackGround(SIP/102-096a48c8, es/ evening ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/evening does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/evening (format 0x8 (alaw)): No suc h file or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/evening -- Executing [...@outbound:19] Hangup(SIP/102-096a48c8, ) in new stack == Spawn extension (outbound, 80, 19) exited non-zero on 'SIP/ 102-096a48c8' This is what the context looks like: [timeofday] exten = 80,1,Answer() exten = 80,n,Verbose( In timeofday ) exten = 80,n,GotoIfTime( 00:00-12:00,*,*,*?day) exten = 80,n,GotoIfTime( 12:01-17:59,*,*,*?afternoon) exten = 80,n,GotoIfTime( 18:00-11:59,*,*,*?night) exten = 80,n(day),Verbose(It's Day..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(Day..) exten = 80,n,BackGround( es/morning ) exten = 80,n,hangup() exten = 80,n(afternoon),Verbose(It's Afternoon..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(afternoon..) exten = 80,n,BackGround( es/afternoon ) exten = 80,n,hangup() exten = 80,n(night),Verbose(Night..) exten = 80,n,BackGround( es/good ) exten = 80,n,BackGround( es/evening ) exten = 80,n,hangup() ___ -- 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] iphone client app
On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote: On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I forgot to mention Ruben's post on this, a review of the apps he has tried. http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod-touch-a-comp At some point I want to take the time to record a call from each of the apps to the same server from the same device and mic (I use an iPod, not iPhone) /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] Dial with timeout don't end call
Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten = 975-INUSE,2,Hangup()
exten = 975-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,us)
exten = 975-NOANSWER,2,Hangup()
exten = 975-NOT_INUSE,1,Dial(SIP/0317998975H323/00733025...@avaya,20)
exten = 975-NOT_INUSE,2,Goto(975-${DIALSTATUS},1)
exten = 975-NOT_INUSE,3,Hangup()
When calling 975, both SIP and cell phone starts to ring.
Answering on the SIP phone, cell phone stop ringing.
Answering on the cell phone, SIP phone keeps ringing.
If not answering any, cell phone stops ringing after 20 sec but
SIP phone just keeps ringing.
== Using UDPTL CoS mark 5
-- Executing [...@inputinterior.se:1] Goto(SIP/0317998985-005e,
975-NOT_INUSE,1) in new stack
--
Goto (inputinterior.se,975-NOT_INUSE,1)
-- Executing [975-not_in...@inputinterior.se:1]
Dial(SIP/0317998985-005e, SIP/0317998975H323/00733025...@avaya,20)
in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- Requested transfer capability: 0x00 - SPEECH
-- Called 00733025...@avaya
-- SIP/0317998975-005f is ringing
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is ringing
-- Nobody picked up in 2 ms
-- Executing [975-not_in...@inputinterior.se:2]
Goto(SIP/0317998985-005e, 975-NOANSWER,1) in new stack
-- Goto (inputinterior.se,975-NOANSWER,1)
-- Executing [975-noans...@inputinterior.se:1]
VoiceMail(SIP/0317998985-005e, 0317998...@inputinterior.se,us) in
new stack
-- Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
-- Playing 'beep.gsm' (language 'se')
-- Recording
the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/inputinterior.se/0317998975/tmp/EKTi4P
format: wav, 0x8c448d0
-- User hung up
== Spawn extension (inputinterior.se, 975-NOANSWER, 1) exited non-zero on
'SIP/0317998985-005e'
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Re: [asterisk-users] iphone client app
I have tried every sip phone offered on Apple's APP store (as of three weeks ago) and the only one that worked well for me was iPico and I think I paid $10.00 or so for it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Sunday, December 13, 2009 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iphone client app On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote: On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I forgot to mention Ruben's post on this, a review of the apps he has tried. http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod- touch-a-comp At some point I want to take the time to record a call from each of the apps to the same server from the same device and mic (I use an iPod, not iPhone) /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 ___ -- 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] iphone client app
Siax is working great for me and as far as I know/remember well, you can get it from the app store for a reasonable price. It supports SIP and IAX2 and works easy with Asterisk. \erik On 13 dec 2009, at 15:32, John Regal wrote: I have tried every sip phone offered on Apple's APP store (as of three weeks ago) and the only one that worked well for me was iPico and I think I paid $10.00 or so for it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Sunday, December 13, 2009 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iphone client app On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote: On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I forgot to mention Ruben's post on this, a review of the apps he has tried. http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod- touch-a-comp At some point I want to take the time to record a call from each of the apps to the same server from the same device and mic (I use an iPod, not iPhone) /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 ___ -- 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] Random DTMF tones generated from speech
Thank you, very interesting! As I understand the Digium card is used as a interrupt source for Asterisk? Is there a diagnostic tool available ? Anybody else experienced a simmialr problem? Thank you! HB From: cov...@ccs.covici.com Date: Sat, 12 Dec 2009 19:04:23 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I used to have this problem with a Digium 400p -- even when not in use and a Motherboard which was inadequate in terms of the interrupts for the Digium card -- when I got a better Motherboard the problem went away. hbk fo...@online.no wrote: Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB ___ -- 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] DEVICE_STATE
Philipp Kempgen wrote: Magnus Benngård schrieb: Set call-limit=10 (or any other value 0) Actually, I believe call-limit is deprecated, and you can instead use callcounter=yes Leif Madsen. ___ -- 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] question on queues
I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry ___ -- 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] question on queues
On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote: I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry In queues.conf, you can have joinempty=no for the selected queue. If there's noone logged in, the dialplan will move forward to the next entry. ---fred ___ -- 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] Asterisk throws error using the alsa module
On Tue, 08 Dec 2009 10:42:58 -0800, Dave Platt dpl...@radagast.org
wrote:
[Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable
I agree, this looks like some form of conflict for the sound device.
i've got so far that i can use pulseaudio normally as asterisk user, but
still not with the asterisk application itself (see below).
aster...@puppy:~$ grep asterisk /etc/group
dialout:x:20:asterisk
audio:x:29:pulse,mpd,vitaminx,asterisk
pulse:x:111:mpd,asterisk
pulse-access:x:112:mpd,asterisk
asterisk:x:114:
The first thing I'd suggest doing, is trying to reproduce the
error with a command-line tool, with asterisk out of the loop
entirely. You'd use a command such as
aplay -D default /path/to/demo-congrats.wav
See if it plays back properly.
Running aplay as asterisk user seems to be no problem:
aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
Little Endian, Rate: 48000 Hz, mono
aster...@puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav
Wiedergabe: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
Little Endian, Rate: 48000 Hz, mono
aster...@puppy:~$ aplay -Ddefault /usr/share/sounds/alsa/Front_Center.wav
Wiedergabe: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
Little Endian, Rate: 48000 Hz, mono
pulseaudio spawns itself as it should:
aster...@puppy:~$ ps -A | grep pulse
23820 ?00:00:00 pulseaudio
this works as defined in /etc/asound.conf:
aster...@puppy:~$ cat /etc/asound.conf
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
i've acticated the alsa plugin for asterisk:
puppy:/etc/asterisk# grep -E 'alsa|oss' modules.conf
load = chan_alsa.so
noload = chan_oss.so
puppy:/etc/asterisk# grep default alsa.conf
input_device=default
output_device=default
A resource temporarily unavailable error from ALSA would tend
to suggest one of two sorts of conflicts:
[1] A low-level (e.g. IRQ) conflict for the sound device itself.
This could occur as a result of motherboard misconfiguration...
for example, if the sound card/chip was configured to use
IRQ 2 or 3, and there was also a serial port in use which
made use of this interrupt. Check (e.g.) /proc/interrupts
to see if you can find such a conflict.
[2] A higher-level conflict for use of the sound card, e.g.
between two different (and incompatible) ALSA accesses,
or between a native ALSA access and a user of ALSA's
OSS driver- or library-level API emulation.
One not-uncommon culprit is having an X Window desktop up and
running. Some of the newer desktop packages have their own
sound-management architecture (e.g. ESD, the Enlightenment
Sound Daemon, or the JACK audio toolkit, or PulseAudio).
These management systems often open the underlying sound
device (in a non-shared mode) and then provide their own APIs
for arbitrating access, mixing multiple outputs together, etc.,
and a separate native ALSA access from Asterisk will often
be unable to share access to the card.
i'm running pulseaudio on top of alsa. through setting /etc/asound.conf as
above any calls to alsa should be redirected to pulseaudio (at least that's
what i thought).
here are some of the relevant packages i have installed:
puppy:/etc/asterisk# dpkg -l | grep -iE 'alsa|asterisk|pulse'
ii alsa-base1.0.21+dfsg-2ALSA driver
configuration files
ii alsa-utils 1.0.21-1 ALSA
utilities
ii asterisk 1:1.6.2.0~rc7-1 Open Source
Private Branch Exchange (PBX)
ii asterisk-config 1:1.6.2.0~rc7-1
Configuration files for Asterisk
ii asterisk-sounds-main 1:1.6.2.0~rc7-1 Core Sound
files for Asterisk (English)
ii libasound2 1.0.21a-1shared
library for ALSA applications
ii libasound2-plugins 1.0.21-3 ALSA library
additional plugins
ii libpulse-browse0 0.9.21-1 PulseAudio
client libraries (zeroconf support)
ii libpulse00.9.21-1 PulseAudio
client libraries
ii libsdl1.2debian-pulseaudio 1.2.13-5 Simple
DirectMedia Layer (with X11 and PulseAudio options)
ii linux-sound-base 1.0.21+dfsg-2base package
for ALSA and OSS sound systems
ii pulseaudio 0.9.21-1 PulseAudio
sound server
ii pulseaudio-module-zeroconf 0.9.21-1 Zeroconf
module for PulseAudio sound server
ii pulseaudio-utils 0.9.21-1 Command line
tools for the PulseAudio sound server
That's what happens when I try to do a test call with asterisk:
aster...@puppy:~$ /usr/sbin/asterisk -c
[asterisk-users] AGI with PHP
Hi All,
I'm having problems getting results from a PHP file. This is what the CLI is
showing.
-- Executing [...@internal:1] AGI(Console/dsp, GoTalk.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php
[Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned
error: Broken pipe
If I run the PHP file from Linux it returns the result I want but how do I get
that result back into Asterisk. I'm using Asterisk 1.6.0.10.
My extensions.conf file looks like:
exten = 111,1,Answer()
exten = 111,n,AGI(GoTalk.php)
exten = 111,n,NoOp(Result is : ${callnum})
exten = 111,n,HangUp()
My php file is:
#!/usr/bin/php
?php
$start = '5';
$mysql_host = 'localhost';
$mysql_user = 'username';
$mysql_password = 'password';
$my_database = 'asteriskcdr';
$dbtable = 'cdr';
$from_date = '2009120500';
$to_date = '20100104235959';
$callnum = '0';
$minutes = '0';
$query_duration = SELECT SUM(billsec) FROM $dbtable WHERE calldate =
$from_date AND calldate = $to_date AND disposition='ANSWERED' AND dst like
'04%';
$query_calls = SELECT COUNT(dst) FROM $dbtable WHERE calldate = $from_date
AND calldate = $to_date AND disposition='ANSWERED' AND dst like '04%';
$query_cb_calls = SELECT COUNT(dst) FROM $dbtable WHERE calldate = $from_date
AND calldate = $to_date AND disposition='ANSWERED' AND dst like '04%' AND src
like '04%';
// Connecting, selecting database
$link = mysql_connect($mysql_host, $mysql_user, $mysql_password)
or die('Could not connect: ' . mysql_error());
mysql_select_db($my_database) or die('Could not select database');
// Performing SQL query
$result = mysql_query($query_calls) or die('Query failed: ' . mysql_error());
// Printing results in HTML
while ($line = mysql_fetch_array($result, MYSQL_ASSOC)) {
foreach ($line as $col_value) {
$callnum = $col_value;
}
}
// Performing SQL query
$result = mysql_query($query_cb_calls) or die('Query failed: ' . mysql_error());
// Printing results in HTML
while ($line = mysql_fetch_array($result, MYSQL_ASSOC)) {
foreach ($line as $col_value) {
$callnum = $callnum + $col_value;}
}
// Performing SQL query
$result = mysql_query($query_duration) or die('Query failed: ' . mysql_error());
// Printing results in HTML
while ($line = mysql_fetch_array($result, MYSQL_ASSOC)) {
foreach ($line as $col_value) {
$minutes = round($col_value/60);
}
}
// Free resultset
mysql_free_result($result);
// Closing connection
mysql_close($link);
echo $callnum,$minutes;
fwrite(STDOUT,$callnum,$minutes);
fflush($stdout);
?
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Re: [asterisk-users] AGI with PHP
On Mon, 14 Dec 2009, David Klaverstyn wrote: I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [...@internal:1] AGI(Console/dsp, GoTalk.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe If I run the PHP file from Linux it returns the result I want but how do I get that result back into Asterisk. I'm using Asterisk 1.6.0.10. Just because your PHP script is executed by the AGI dialplan application does not make it an AGI. Your script does not follow the AGI protocol. It does not read the AGI environment, it executes no AGI requests, and it reads no AGI responses. You should do some reading on what an AGI is. http://www.voip-info.org/wiki/view/Asterisk+AGI would be a good start. 2 (biased) suggestions: 1) Use an established AGI library. Nobody gets it right the first time. 2) Consider using a compiled language like C. You can execute xxx's of AGIs written in C in the time it takes to load PHP and parse your script. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] iphone client app
On Sun, Dec 13, 2009 at 8:45 PM, meetmecall i...@meetmecall.nl wrote: Siax is working great for me and as far as I know/remember well, you can get it from the app store for a reasonable price. It supports SIP and IAX2 and works easy with Asterisk. It looks like it requires a jailbroken iPhone, am I wrong? /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
Re: [asterisk-users] question on queues
Hi Jerry, I use the built-in function queue_member http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member?type=functionsvalue=QUEUE_MEMBER and check with a GotoIf statement to check if the number is equal to zero. If it is not I send the call to the queue, if it is I pass the call to dial a cell-phone number or go directly to voicemail depending on which queue the call was originally destined for. Travis - Original Message - From: Jerry Geis ge...@pagestation.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 13, 2009 4:20:40 PM Subject: [asterisk-users] question on queues I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry ___ -- 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] iphone client app
Fring, it's free and works perfectly with an Asterisk server.. On 13 Dec 2009, at 10:15, Alex Samad wrote: Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex ___ -- 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
