[Asterisk-Users] Falsh Panel in Xorcom Rapid
I have a clean install of rapid 1.1 installed. I have installed the Flash Operator Panel from the Install Other Software menu. I am able to log into the panel from another computer on my network but all I see is the Conference Room 300. There are no extensions or any other options on the panel. Clearly I am missing something. Can anyone giude me as to how to get the Panel going so that I can see the extensions and other options. Thanks in advance. Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] OT: Broadvoice is finally starting to giveanswers
Isn't Global Crossing Based in Bermuda? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, May 12, 2005 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Broadvoice is finally starting to giveanswers Undisputed charges left unpaid during the dispute is not grounds to cancel service in america. Thus while the undisputed charges are paid broadvoice cannot claim that BV was behind on payments to terminate service. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 5/12/2005 ___ 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
RE: [Asterisk-Users] unlimited iax termination
Broadvoice is $20/unlimited in/out to USA, Canada and Western Europe Myphonecompany is $24.95/mnth in/out Unlimited to USA Dialpad offers unlimited outgoing for $11.99/month I agree that IAX2 is a better protocol but I personally use all three of the above and the quality is adequate for home use. On the other hand, I use IAX2 with [EMAIL PROTECTED]/min for my business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:02 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] unlimited iax termination Can you tell me, which SIP providers that *work with asterisk* are providing unlimited inbound/outbound calling for $20? Iconnecthere is 30, broadvoice is 30, etc..., and will allow you to specify your outbound cid? Seriously, I'd like to know, the whole point of this is market research, we want to know what people want to buy and what they want to pay for it. Also, IAX2 is, imho, a better protocol than SIP over udp especially for users with standard broadband connections to significantly oversubscribed ISPs, but I'm sure that will spark some discussion... -Joe --- Mike Matthews mike74105 at cox.net wrote: Why would anyone pay $19.95/mth for termination only when you can pay the same price and get unlimited INCOMING (DID) and outgoing with numerous SIP providers? -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of joe at jsci.net Sent: Thursday, April 07, 2005 10:07 AM To: asterisk-users at lists.digium.com Subject: [Asterisk-Users] unlimited iax termination ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 4/7/2005 ___ 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
RE: [Asterisk-Users] unlimited iax termination
Why would anyone pay $19.95/mth for termination only when you can pay the same price and get unlimited INCOMING (DID) and outgoing with numerous SIP providers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 07, 2005 10:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] unlimited iax termination We are planning on offering unlimited IAX terminations to the US for residential/home users for USD $19.95 per month, and business users for a higher price (not yet determined), starting May 1st, but we wanted to see what kind of interest there will be for this first. If you might be interested in this, please send an e-mail to [EMAIL PROTECTED] with contact information. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 4/6/2005 ___ 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
Re: [Asterisk-Users] Asterisk@Home Question
Have each phone peer reference a different context. Have the different contexts set up to use the particular service for that user. - Original Message - From: * KAPIL * [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 11:52 PM Subject: [Asterisk-Users] [EMAIL PROTECTED] Question Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Any ideas? ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005 ___ 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
Re: [Asterisk-Users] broadvoice
They charge you by the minute for the second and more concurrent calls if you are on an unlimited plan. - Original Message - From: JD Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 1:41 PM Subject: Re: [Asterisk-Users] broadvoice Im curious about that too.. if so how many concurrent calls will they allow? JD Matt wrote: Hi, I'm currently routing my asterisk server out over broadvoice.. it seems I can do multiple outgoing and incoming calls does anyone know if broadvoice actually allows this or not? ___ 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 -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005 ___ 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
Re: [Asterisk-Users] Preserve g729 registration over reinstall??
For the record here, I am quoting from an email from Digium on the subject: "You will need to backup /var/lib/asterisk/licenses. You will also needto backup the codec_g729 and format_g729 in your/usr/lib/asterisk/modules/ directory. The ethernet cards in yourmachine cannot be changed. Otherwise you will have to reregister yourcodec. If required you may reregister your codec. If you run into any problemsreregistering, we will assist you on with that problem. Please refer to http://www.digium.com/index.php?menu=asterisk_g729 foradditional instructions." Thanks to Digium Support for the prompt and thorough response.Mike Matthews [EMAIL PROTECTED] wrote: I purchased the g729 codec from Digium. Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall? Perhaps by backing up a directory or a file? Any help appreciated.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users"Put down that coffee...coffee is for Closers!"Phone: 918-770-4503Fax: 206-666-1720email: [EMAIL PROTECTED]sip: [EMAIL PROTECTED]___ 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
[Asterisk-Users] Preserve g729 registration over reinstall??
I purchased the g729 codec from Digium. Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall? Perhaps by backing up a directory or a file? Any help appreciated.___ 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
Re: [Asterisk-Users] g729 Lic ordered from Digium Question.
You will probably get it Monday morning. Thats what happened to me when I ordered on a Friday noght. - Original Message - From: David Uzzell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 13, 2005 8:10 PM Subject: [Asterisk-Users] g729 Lic ordered from Digium Question. Does anyone know how long the orders take? I ordered some a couple of days ago and it said normally 24hours, and I am guessing that the weekend cause's some delays but it did not say anything abouy that. Any one got any ideas on how long generally over the weekend it takes? Thanks David ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.2 - Release Date: 3/11/2005 ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ 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
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
Thanks. Adding those lines appears to have fixed the problem. I'll just hold on til the NEXT TIME Broadvoice decides to make a change. Thanks again. Bartosz Wegrzyn - asterisk wrote: I don't know what is wrong with the Broadvoice, but for me everything works fine. I used the setup they provided on their website. It works fine and with no problems. To make sure that all incoming calls will never miss my box I added those lines in sip.conf. For me it works fine. [broadvoice-incoming] type=peer host=147.135.8.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming2] type=peer host=147.135.0.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming3] type=peer host=147.135.4.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never This helps in case of any dns problems with resolving sip.broadvoice.com Bart, ___ 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
Re: [Asterisk-Users] LiveVoIP Problems?
I have to say that I have had excellent service and support from Livevoip as well. Every support issue I have had has been handled inside of 2 hours and that includes weekends. I have had a Sr. VP there call me at midnight on a Saturday night to work out a problem. They have DID's in MANY areas that you won't see elsewhere and buying one is so simple it's childs playjust pick the number you want and click. I have recommended them to others and until they screw up big time, I will continue to use them myself. Of course, all this is moot if you need Asterisk IVR ring-back :) Steven Frazier wrote: I have about 10 DIDs, I had an issue that lasted a day or so that was Level 3's issue, it took about 12 seconds before the calls would come in. That was resolved and I haven't had any issues at all. I appreciate the fact that there are reselling Level 3 DIDs since they seem to be in a lot of cities, towns and now some burgs. I have had excellent support response anytime that I have had issues. I have talked to several folks there when the issue of the long wait for the call to complete. I also had a call that advised me that the ringback issue appeared to be with asterisk. I explained to them, like others have, that the issue is only with LiveVOIP not with other providers that I also have like TXLINK, Teliax, NuFone. I am not a programmer, but they advised me that they problem didn't reside in SIP. So, I took a chance to try it. I ordered yet another DID with SIP vs IAX. No ringback issue on the SIP configured DID at all. ___ 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
Re: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVSHEAD - VERIFIED
This works for me both incoming and outgoing w/Sipphone. Note there is NO username, secret entries in the peer definition. I am using * vers 1.05 register=1747nnn:[EMAIL PROTECTED]/1747xxx ; note:extension in extensions.conf matches for incoming [sipphone] type=peer host=proxy01.sipphone.com dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc fromdomain=proxy01.sipphone.com qualify=no reinvite=no insecure=very nat=yes OK, here is the sip.conf entry: register=1747XXX:[EMAIL PROTECTED]/4321 [proxy01.sipphone.com] type=peer ;auth=md5 secret=YYY username=1747XXX fromuser=1747XXX fromdomain=proxy01.sipphone.com host=proxy01.sipphone.com nat=no qualify=no canreinvite=no disallow=all allow=ulaw ;context=default ;callerid=Hadar Pedhazur 1747XXX ___ 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
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
Why can't Broadvoice just LEAVE WELL ENOUGH ALONE!! Now, after applying these new variables, I can't receive INCOMING calls. Sheesh, what a bunch of BS!! Now we have to spend another weekend fixing what BV screws up. Dan Weber wrote: Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ 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
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
EXCUSE ME!! I changed NOTHING except added the variables you indicated. Then incoming calls stop. So I change back to prior sip.conf and incoming calls work again. So you tell meif they are totally unrelated, then why do incoming calls go straight to BV voicemail when I apply your changes and start working again when I remove your changes? Also, insulting your customers is not the way to keep them. Or maybe BV wants to get rid of Asterisk users. Dan Weber wrote: They are completely unrelated. Maybe you should read instructions. Dan On Sat, 5 Mar 2005, Mike Matthews wrote: Why can't Broadvoice just LEAVE WELL ENOUGH ALONE!! Now, after applying these new variables, I can't receive INCOMING calls. Sheesh, what a bunch of BS!! Now we have to spend another weekend fixing what BV screws up. Dan Weber wrote: Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ 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
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
Thanks for sending this. However, I literally cut and pasted your examples (with my sip credentials) and incoming calls still go automatically to BV Voicemail. Using sip debug shows that the call never hits my * box. Thank anyway...it was certainly worth a try. Marios Andreou wrote: Its working just fine for me. All IN and OUT. sip.conf: register = [EMAIL PROTECTED]:PP:[EMAIL PROTECTED]/ext Where PPP is the password in your Account and not the login password for BroadVoice. ext is the extension to ring make sure that it is registered again with * once you restart it. Then: [broadvoice] type=friend username=XX fromuser=XX fromdomain=sip.broadvoice.com secret=PP host=sip.broadvoice.com port=5060 dtmfmode=inband insecure=very context=broadvoice qualify=yes disallow=all allow=ulaw ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no That's it. ___ 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
[Asterisk-Users] Xten needs beta testers for Linux X-Lite Softphone
X-Ten is beta tesing a new softphone for Linux and could use some beta testers. If interested, please contact: Neil McGuigan, [EMAIL PROTECTED] I am currently testing it and it appears every bit as good as the Windows version. ___ 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
Re: [Asterisk-Users] Xten needs beta testers for Linux X-Lite Softphone
It appears that it just supports SIP. [EMAIL PROTECTED] wrote: snip X-Ten is beta tesing a new softphone for Linux and could use some beta testers. If interested, please contact: Neil McGuigan, [EMAIL PROTECTED] I am currently testing it and it appears every bit as good as the Windows version. /snip Will / does it support IAX2 or just SIP? -Ron ___ 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 ___ 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
