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Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- Chris

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

I just wanted to send out some information that will hopefully help others. I don't know, maybe I'm the only one that's been having problems with this. I've been pulling my hair out for a while wondering why Google would not send my incoming calls to my Asterisk box. The calls would just roll to voice mail and no packets ever reached Asterisk. This has happened on two separate

After an upgrade to asterisk 1.6.2.1 I'm unable to make outgoing calls via Vitelity. I get lots of these on my asterisk console: [Jan 27 08:58:41] WARNING[25653]: chan_sip.c:3581 __sip_xmit: sip_xmit of 0x834ae08 (len 927) to 64.2.142.18:0 returned -1: Address family not supported by protocol There's a bug report that seems to address this but it's categorized as minor and

I know that Asterisk can use the system's sound card as the output device for a console channel. However, I'm using Asterisk call files and would like to be able to hear the calls over a set of speakers as the call files are being processed. Basically I'm wanting to listen in on the calls as they happen. Right now everything is happening silently in the background. Can a sound card

I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have callprogress=yes in chan_dahdi.conf because, from everything I've read, it is needed when using call files over PSTN, which I DO use occasionally. I know that callprogress=yes is "experimental" and causes some issues. We've never experienced any problems when making local calls over PSTN with callprogress

When I call into my Asterisk box via my VoIP line (using gsm codec) and then try to make an outgoing DISA call over PSTN I get the following: [Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot handle frames in gsm format [Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to forward voice or dtmf Obviously, it looks like asterisk is not converting the

I've been playing with call parking in Asterisk 1.8.1. I'm able to park a call and then pick it back up. However, on the second attempt, the #72 DTMF is ignored. Asterisk just passes that DTMF on to the caller and the call parking never happens. Shouldn't I be able to park a call more than once? -- Chris -------------- next part -------------- An HTML attachment was scrubbed...

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? -- Chris -------------- next part

Hello. I would like to configure Asterisk to accept incoming calls from two different GoogleVoice numbers via gtalk and jabber. I'm running Asterisk 1.8.3.2 and I can get one number working just fine. However, I can't figure out how to modify the gtalk.conf file shown on the Asterisk wiki site to work with two different jabber profiles. Do all incoming GoogleVoice calls have to go

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110224/2f0d56bc/attachment.htm>

OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand