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...Friday, August 13, 2004 9:32 AM Subject: RE: [Asterisk-Users] Broadvoice User hung up on voicemail > I place a call through Broadvoice to a phone and put it on mute(no > noise) and it didn't get disconnected. > > > -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: Friday, August 13, 2004 12:23 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail > > Ok, here's an update... I tried also just using the Record() application > and > the same thing seems to happen... It...

...with 3Mbit downlink and 256Kbit uplink... As I said earlier I wouldn't have even asked, but it dosen't seem to be totally bandwidth related so I'm wondering if anyone has any ideas... Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris

...ays it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs ove...

..._______ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ From: "Chris Shaw" <chriss@watertech.com> mmmmm I can see it now.... The linux kernel having to ask permission from the windows kernel to take over an interrupt... but the windows kernel is too bogged down with a buffer overflow exploit or sending spam to everyone in the universe... Or worse! You're in the middle of a call and...

I have a bridge that has gigabit interfaces. The machine in question has the fun job of being a Bridge, Firewall and SMB server. Both of the Gigabit interfaces are connected to workstations directly via Xover cable (well MDI-X to be exact). My question is, if I enable jumbo frames on the gigabit interfaces will that make any difference in overall transfer rate of the bridge? I was thinking it

Did my last post make sense? Is this a known issue with the bridge-nf code? Is there something I can do to help? Should I just shut up now? -Thanks in advance Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not saving to voicemail.conf. So new passwords are lost when asterisk is restarted and people

...oks kinda nice, where can one get hold of it? How about it's * compatibility? I realize that it says it does things like 3-way conference and attended transfers, but how about in *? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

...ansfer, the far end ringing sounds terrible and even skips a few rings... If I remove the macros and put things back the way they were, things return to normal... This is VERY wierd... -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

...tomorrow. If you are > interested I can send you a diff (stable version 0.9.1). As promised yesterday: Anybody interrested can download the patch for Asterisk 0.9.1 at http://matthiasendler.net/asterisk/patch/. Best regards Matt --__--__-- Message: 8 From: "Chris Shaw" <chriss@watertech.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] D-Link DPH-80S vs * Date: Thu, 22 Jul 2004 11:37:18 -0700 Reply-To: asterisk-users@lists.digium.com What about the DPH-1008 or is that just the US version?? -Chris ----- Original Message ----- From: "Kanur...

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

...oks will be in gaim to interface it. > Maybe like a iaxtel/* protocol plugin. I'm still speculating about > details though. I've got most of the lower stuff done now. > > Matthew Fredrickson > > --__--__-- > > Message: 9 > From: "Chris Shaw" <chriss@watertech.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] BroadVoice problems? > Date: Mon, 19 Jul 2004 08:43:07 -0700 > Reply-To: asterisk-users@lists.digium.com > > Now that you mention it, yes... it seems that SIP isn't being passed from > the...

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

...nbsp;</DIV> > <DIV><FONT face=3DArial size=3D2>Any ideas?&nbsp; = > </FONT></DIV></BODY></HTML> > > ------=_NextPart_000_02C1_01C45072.02CC3E10-- > > > --__--__-- > > Message: 2 > From: "Chris Shaw" <chriss@watertech.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] feature - VM gain adjust? > Date: Mon, 12 Jul 2004 08:43:19 -0700 > Reply-To: asterisk-users@lists.digium.com > > Hmmm... I don't know if playing with the * code would really be the best &...