similar to: hdmi an console dsp

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

I am having a problem with SpanDSP. What happens is when I send a fax to SpanDSP the fax message seems to fail in the training phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

I downloaded spandsp0.0.6pre17 I download http://sf.net/projects/agx-ast-addons for app_txfax and found trunk/app_fax to be newer so I used that. spandsp compiled fine. app_fax compiled when loading I get: [Apr 5 08:55:54] ^[[1;31;40mWARNING^[[0;37;40m[7505]: ^[[1;37;40mloader.c^[[0;37;40m:^[[1;37;40m433^[[0;37;40m ^[[1;37;40mload_dynamic_module^[[0;37;40m: Error loading module

>> 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, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load => chan_alsa.so noload => chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

Hi: We just setup the Asterisk and it seems to start ok. We checked the log, and beside the timer warning, there isn't other error message. However, we tried both SIPURA and XLite, but their registration is not accepted (timed out and failed). Could someone tell me what's wrong? [message] Dec 10 01:33:22 WARNING[2649]: Unable to open IAX timing interface: Permission denied Dec 10

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a "conference" room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone giving a speech but not necessarily interact. I've got confbridge configured and

Is it possible ot have multiple contexts in alsa.conf [general] autoanswer=yes input_device=hw:0,0 output_device=hw:0,0 [usbaudio] autoanswer=yes input_device=hw:1,0 output_device=hw:1,0 is Dial(Console/usbaudio) valid? I presume Dial(Console/dsp) uses the default or general context above. how do you direct a call to the second ALSA device usbaudio? Thanks, jerry

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

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Currently I am having 2 issues with my Avaya 4602 phone: First, the phone registers with my Asterisk server, but when I start dialing I get a busy signal after 4 digits. I specified in the dialplan on the phone to expect 10 digits and that solved that problem, but I still immediately get a busy after the 10th digit. The phone never sends a dial command to asterisk. Second, asterisk is

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the