similar to: GrandStream BT101 Attended Transfers

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

Back to the ACTUAL TOPIC of this thread... This phone looks 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

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.. this is really wierd... I just cleaned up my dialplan a bit by adding some macros with a strange side effect... On my incoming context which has no macros in it, far end ringing used to work... now that I have macros defined, far end ringing has stopped working all together... The macros DO work, but when they transfer, the far end ringing sounds terrible and even skips a few rings... If I

> -----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 over TCP and TCP >

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

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

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

Now that most of you have worked overtime to show why most people are continually pissed at Nix Users (all except two of course). The problem I can see is the downright technosnobbery involved. There is nothing wrong with Linux. I play around with RH9 and FreeBSD and find that most things run fine. But you get into a problem where it keeps asking for the same blamed libraries over and over on

Hello, Has anyone noticed that if you don't pick up a BT101 phone in 60 seconds it stops ringing and acts like it was never called ? Or is it just something I missed ? If it matters for something I have call waiting enabled on the phone. Product Model: BT100 Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6 Custom Ring Tone:

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

Hi, I am wondering if anyone has gotten the BT101's to work with the paging in Asterisk? I know that the phones themselves have an auto-answer option and if I turn it on every call is auto answered. I want to be able to call the extension normally and have it ring normally but if someone dials # and the extension to have it auto answer for intercom purposes. Anyone have this working?

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

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've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

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

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

Got one weird one and one prob easy one. 1. I have upgraded our BT101's to Program--1.0.5.0 Bootloader--1.0.0.17 HTML--1.0.0.34 VOC--1.0.0.6 after doing this i have some phones on different subnet's ie 255.255.255.248 or .192 or .252 and i am now unable to login to these phones from different subnet's . I have one at home which is on a .248 ( Using an external IP for the phone )