similar to: Asterisk 11 and pulseaudio setup as local user

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

alsa_card_init^[[0m: snd_pcm_open failed: Connection refused soundcard_init^[[0m: Problem opening alsa capture device These are the errors I get. I changed the following: chown -R myuser:myuser /var/log/asterisk chown -R myuser:myuser /var/lib/asterisk chown -R myuser:myuser /var/spool/asterisk chown -R myuser:myuser /var/run/asterisk chown -R myuser:myuser /usr/lib/asterisk edited

Hi, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1

asterisk-1.4.7, Fedora 7, intel emt64 - nocona: == Parsing '/etc/asterisk/alsa.conf': Found ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to open slave [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365 alsa_card_init: snd_pcm_open failed: No such file or directory [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:481 soundcard_init: Problem opening alsa I/O devices == No sound

I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir => /var/run/asterisk runuser = asterisk rungroup = asterisk The directory,

Hi! I am interisted to dial out from the console with chan_alsa. Can somebody of you help me according this problem?! I added user the asterisk to "pulse" and "pulse-access", and it didn't change anything. alsa applications are routed by default to pulse. cat /etc/asound.conf pcm.!default { type pulse } ctl.!default { type pulse } What might be the problem?!

Ok digging deaper... I was always trying to run the session as su myuser -c "asterisk -fn" This does not seem to work. If I login as myuser and run "asterisk fn" it worked... I got a lot of crackly noise that I normally dont have but it worked. Any thoughts on why I cannot run the command as 'su myuser -c "asterisk -fn"' ? Thanks jerry -------------- next

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Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at 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 > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

Ah, no, you misunderstand. Asterisk wouldn't care one little bit what is on the page - Chromevox would do all that. A screenreader usually tabs or arrows their way about, selecting headings to read content. Thus, Asterisk ONLY needs to be able to hear content FROM the browser and pipe it to the channel, and pass keypresses back TO the browser. The human is the parser, if that makes sense?

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few