similar to: asterisk-users Digest, Vol 68, Issue 4

Hi, I need some help debugging a sip situation. I started to have problems with sip trunks, using more than one trunk (and sometimes using only one) the sip module seems to freeze. My local extensions lost registration and also the trunks. The only way that I can restart the sip is removing the trunks. If I make sip reload or restart asterisk the sip module takes many many time before

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> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

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Hi, I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP provider via internet. I noticed Asterisk gets slow and behaves in strange manner if I unplug my internet cable from the PBX: for example I get incoming calls after seconds or I get no audio during calls. I thought it was something connected to DNS resolution so I put VoIP provider addresses inside /etc/hosts but

Just for the sake of this thread I'll paste part of the last post regarding this issue in the asterisk bug tracker. kpfleming on 2005-03-10 post: "Essentially, what we are saying is that if you are going to use DNS to resolve critical information in your Asterisk configuration, you need to do everything possible to ensure that the DNS lookups will not block for long periods of time.

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On 12/17/2013 07:00 PM, gluster-users-request at gluster.org wrote: > Contents of

>> Is there a defined format tag for Opus that should be used with the >> WAVEFORMATEX structure? See wFormatTag >> (http://msdn.microsoft.com/en-us/library/windows/desktop/dd390970.aspx). > > There isn't one. Putting Opus into RIFF is complicated by the fact that > it requires both variable-duration and variable-size frames, though I > suppose something could be

Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45

yes you can easily find format convertors, check on internet there are so much options to select. you can easily swing between different formats. check on internet wave to mp3 convertor or DSS to wave convertor. On Fri, Jan 8, 2010 at 1:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or

Nicolas, Thank you for your response. I had tried that before and it didn't work. I am trying to look up the route for a dialed number, so its a full E.164 number. Please see my query below when I try to look up the route for a USA number; mysql> SELECT * FROM routes WHERE "^13237309880" RLIKE pattern ORDER BY LENGTH(pattern) DESC;

> Message: 13 > Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST) > From: =?iso-8859-1?q?Eugen=20Cristea?= <tecristea@yahoo.co.nz> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] asterisk to asterisk config > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I would like to set two separate asterisks to talk to > each other. > Any

>> >> >>Can I program a specific C.O. line directly to a button? >> >> Adopting the Critchfield style for a moment, no, *you* probably can't. But, depending on one's expertise with the hardware and Asterisk, it can certainly be done. I am by no means an Asterisk expert but have about 35 years of random phone experience. I have an ancient Siemens HiNet

okay, my needs have been met. Thanks much. Now, for the sake of Google, could someone respond with how Asterisk likes single vs. HT vs. dual core vs. dual cpu, etc.? --Mike > ------------------------------ > > Message: 15 > Date: Sun, 08 Jan 2006 18:57:12 -0800 > From: Mike Fedyk <mfedyk@mikefedyk.com> > Subject: Re: [Asterisk-Users] Processor Update? > To:

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If i change the configuration and i send the call to an internal sip

> Letting a carrier use you as a carrier seems like quite a bad idea generally.. I think I would agree. :) > > _NXXNXXXXXX => Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here get routed upstream.... > in the 'local' context instead of the other one?.... > So here is where the finer points of Asterisk pattern matching must come into play. All of the

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Hi Joseph, With iax servers dispersed across the internet, you could still use the below setup, it would work but it's not as secure as you would want it. I would then have a context for each server and use the IP address deny and permit statements. Also, you can have 1 server with a public IP and have the other servers behind a NAT register to the public server. There are really several