search for: punknow

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

I know someone who's at VON this week. Apparently Mark Spencer was up there talking about how Asterisk supports SRV. Sounds like vaporware to me. > -----Original Message----- > From: David Thomas [mailto:punknow@gmail.com] > Sent: Thursday, March 16, 2006 11:54 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and > Polycomphone!! > > > In regards to HA... > > SER is definitely a good option, b...

...8.12:5060:3600:3254104:sip:3254104@216.188.128.12 As you can see, that isn't what the REFER has. It has 216.188.140.203, which is Asterisks IP address. I don't know if that's the issue or not. Asterisk _IS_ in the RTP path. Doug. -----Original Message----- From: David Thomas [mailto:punknow@gmail.com] Sent: Friday, March 03, 2006 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes Sorry, I saw that right after I posted. It is per month. And almost all during business hours. regards, David On 3/3/06...

...Mail. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/c39f88 17/attachment.html ------------------------------ Message: 15 Date: Thu, 16 Mar 2006 11:53:55 -0700 From: "David Thomas" <punknow@gmail.com> Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!! To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <27cfba050603161053vdb2d88eqe7dd9327179e1feb@mail.gmail.com> Cont...

Hello, does anyone else have a problem with Asterisk crashing right after a valid password/PIN is entered when trying to access voicemail in the 1.4b3 version? Not sure if this is anything to do with "realtime" per se but I keep getting the asterisk process bail on me as soon as a valid PIN is entered. Anyone? Cheers Ranj

Just wondering if there is much CPU overhead in the translation from IAX2 to SIP, and how taxing this function is as compared to transcoding. We're trying to build an efficient system and would like to avoid taxing the CPU as much as possible. Our upstream service provider is 100% SIP, however we'd like to use IAX2 in our network as well, if it does not cause too much overhead. Not sure

Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

> I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. > All seems to work normally with realtime voicemail, reads vmbox > parameters from the db fine. When I try to change the password, > asterisk operates normally, "enter new password" ok, "re-enter new > password" ok, "password has been changed" > > There are no entries in

Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

...The first question. (Doug Lytle) > 25. Re: Re: Is asterisk able to integrate with MS SQL (Sharon Lim) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 14 Nov 2006 12:00:25 -0700 > From: "David Thomas" <punknow@gmail.com> > Subject: Re: [asterisk-users] Load balance Asterisk servers? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <27cfba050611141100u4793e56dkdecd5c8a786f4625@mail.gmail.com> >...

Great Email. I'm going to respond to some of the points. "Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now." That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters

I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling - Calls come in to asterisk via SIP/IAX and terminated to third party provider via SIP - Codec usage will be about 70% g711 & 30% g729 (there should be no transcoding) - 100% IP setup with no voice cards in the box They have a box on hand with a single 3.2ghz P4 w/Hyper-threading,

On 11/8/06, Thomas Blanchin <tblanchin@gmail.com> wrote: > Hi David. > > I read your post on : > http://lists.digium.com/pipermail/asterisk-users/2006-September/167456.html > > I am in the same situation as you are. I'm looking for a way to > monitor iax2 connexions on asterisk. I'm using sipsak for sip > connexions. > I'm looking for a very simple

We use only IP connections to our asterisk boxes. Given this our origination/termination providers usually send/receive traffic to/from our network on a single IP or limited number of IPs. In a DUNDi Asterisk Cluster, would each of the boxes need to be able to connect to our origination/termination providers directly, or would we need to setup a common gateway box to forward calls to/from our

Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David

I am just wondering if someone can explain the difference between Packek2Packet Bridging vs. Native Bridging in Asterisk. I'm basically tyring to make sure the media travels end-to-end and I've see both of these bridging types mentioned on the asterisk console. Regards, David