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
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=functions&value=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" To: "Asterisk Users Mailing List - Non-Commercial Discussion" 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
On Sun, Dec 13, 2009 at 8:45 PM, meetmecall 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] 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
[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 = $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($stdout); ?> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk throws error using the alsa module
On Tue, 08 Dec 2009 10:42:58 -0800, Dave Platt wrote: >> [Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error: > Resource temporarily unavailable > > I agree, this looks like some form of conflict for the sound device. > i've got so far that i can use pulseaudio normally as asterisk user, but still not with the asterisk application itself (see below). aster...@puppy:~$ 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:~$
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
[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] 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
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 > > To: > Asterisk Users Mailing List - Non-Commercial Discussion > > > 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 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] 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 > wrote: >> On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad >> 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] 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 $10.00 or so for it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Sunday, December 13, 2009 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iphone client app On Sun, Dec 13, 2009 at 11:24 AM, Randy R wrote: > On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad wrote: >> Got a new iphone, want to know about peoples experience with any apps >> that work well with asterisk and run on a iphone > > http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I forgot to mention Ruben's post on this, a review of the apps he has tried. http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod- touch-a-comp At some point I want to take the time to record a call from each of the apps to the same server from the same device and mic (I use an iPod, not iPhone) /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial with timeout don't end call
Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${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/0317998975&H323/00733025...@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/0317998975&H323/00733025...@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' ___ -- 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 wrote: > On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad 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
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 wrote: > From: Warren Selby > Subject: Re: [asterisk-users] Unable to open file... > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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 > > 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
-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 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] iphone client app
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad 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
[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
[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? ___ -- 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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] 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