Re: [Asterisk-Users] USB headsets?
I personally use the Logitech USB Headset 250, been real happy with it. Works great with EyeBeam, X-Lite and Diax so far.. Hope that helps.. -- VoIP Street Origination Services http://www.VoIPstreet.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, May 24, 2006 3:01 AM Subject: [Asterisk-Users] USB headsets? Hi, What USB headset would you recomend? We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Asterisk Problem
Have you configured your extconfig.conf file? If so, do you have lines similar to this: sipusers = mysql,database_name,table_name sippeers = mysql,database_name,table_name Of course you also have to have the tables correctly setup on the MySQL server. Hope that helps.. -- VoIP Street Origination Services http://www.VoIPstreet.com - Original Message - From: Chandan Mishra [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, May 24, 2006 6:40 PM Subject: [Asterisk-Users] Realtime Asterisk Problem Hi i am using the asterisk server on one machine and mysql on another machine.I have my mysql running on 192.168.77.75 and asterisk running on the 192.168.77.77. when executing following cli command on asterisk server on 192.168.77.77 *CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username root for 8 minutes, 45 seconds. But my phones are not getting registered with the 192.168.77.77. The phones have entry in the mysql database. my res_msql.conf on machine 192.168.77.77 have the following entries [general] dbhost=192.168.77.75 dbname=asterisk dbuser=root dbpass=root dbport=3306 dbsock=/tmp/mysql.sock When the cli shows the asterisk connected then why my phones are not able to register the 77 asterisk server. If i am running asterisk on 75 itself then the phones are able to register with the 75 server (where mysql is installed). Please suggest the solution. Thanks Chandan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP re-invite and billing
I know this may sound like a stupid question but I will put on my flame retardant suit and ask anyway. Is there any way to use/allow SIP reinvite and still track the length of the call? I realize that the whole idea of reinvite is that it takes the proxy out of the media path which, from what I understand also kills the proxy's ability to track the start/end time of the call for billing purposes. Are there any really smart guys out there with propeller hats that have come up with a way to get the best of both worlds? Do we lose anything else using reinvite with Asterisk? Thanks in advance for any help.. --Mojo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between VoiceMail and VoiceMail2?
Can someone explain the difference between VoiceMail and VoiceMail2? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dlink DI-102 QOS Thingy?
Anyone using one of these as a QOS device in an Asterisk environment? If so, does it work well? Do you know what exactly it prioritizes? SIP only? IAX? I bought one to play around with but read that it also prioritizes streaming media in general.. The last thing I want is for this thing to give priority to someone who is streaming video and squash the phone calls just so the video looks good. I don't think this thing is going to work as I hoped (a simple/cheap device that will give priority to SIP and IAX). Thoughts? Here is the link: http://support.dlink.com/products/view.asp?productid=DI%2D102 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Channel Banks, what are they for?
Trying again, this never made it to the list for some reason. Thanks.. - Original Message - From: Mojo Jojo [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Thursday, September 29, 2005 10:26 PM Subject: Channel Banks, what are they for? Can someone explain to me what a channel bank is used for? For example, if I had a 24 channel PRI setup with an Asterisk box attached to it via a TE110P, how would a channel bank make my life better? Anyhow, I just have not clue and thought someone could tell me the purpose of these devices. Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com ___ --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] Fw: Channel Banks, what are they for?
Kind of the idea I got.. So then I could take a channel bank and feed my PRI from the telco into it then on the other side I would have a bunch of RJ11 ports to use as normal analog type phone lines? What about this situation... I have a PRI going directly into an Asterisk box now in a location using a TE110P card. What if I wanted to pull off a few of those 24 channels to use for fax lines, dialup internet testing lines etc.. Would this be a job for a channel bank? If so, could you explain how a channel bank, and my Asterisk box would be setup to do something like this? --Todd - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 02, 2005 8:31 AM Subject: Re: [Asterisk-Users] Fw: Channel Banks, what are they for? On Sunday 02 October 2005 09:09, Mojo Jojo wrote: Can someone explain to me what a channel bank is used for? For example, if I had a 24 channel PRI setup with an Asterisk box attached to it via a TE110P, how would a channel bank make my life better? Anyhow, I just have not clue and thought someone could tell me the purpose of these devices. A channel bank is a device used to aggregate (or break apart) individual FXS and FXO ports to/from a T1 connection. Basically, channel banks save time, energy, space and wiring mess. They also often give higher FXO/FXS interface quality as well. They cost a little more but IMO if you take everything into account they are by far cheaper in the long run. Anyway as mentioned channel banks are an easy/cheaper way to aggregate lines. Instead of having 24 Sipura boxes or 6 TDM400P cards and all the wiring mess that goes with it, you have a single T1 card and a single cable going to a single channel bank, which will have a single D50 connector that you can terminate directly to a single BIX punchdown strip. Channel banks are also typically (much) higher quality than the cards or low-density ATAs you can buy. I am particuarly fond of Carrier Access Adit600s. -A. ___ --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 ___ --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
[Asterisk-Users] IP Cop as a firewall and QOS
We are looking for a good firewall replacement which will basically do pot blocking and QOS. Our current solution just plain stinks.. We basically need to handle the traffic of a few web servers, mail server and asterisk box. The most traffic this device will need to handle is what can be shoved through a T1. I don't mind buying an appliance to get something solid but IP Cop just looks better than he appliances I see out there. I am only concerned if it is stable for a production environment. It says it's designed for a SOHO environment, we are doing a bit more than that. Will this thing hold up? Can it be trusted? Anyone using this for QOS and Asterisk in a production setup. Any thoughts or suggestions or warnings would be appreciated! Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.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
[Asterisk-Users] 10 digit dialing in Ft Lauderdale, FL?
Does anyone know if 10 digit dialing is used in Ft Lauderdale, FL? -- Private Label Wholesale Internet Access! http://www.YourOwnISP.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
[Asterisk-Users] Wiki Trouble?
Anyone having trouble getting to the Wiki? http://www.voip-info.org/wiki-Asterisk -- Private Label Wholesale Internet Access! http://www.YourOwnISP.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] Pattern Matching
I do this already with outgoing calls and it works fine as long as I am only using the Dial command. Where I am running into trouble is when doing something like I have created below. I know the syntax is not 100% correct, just using it as a quicky example. What happens here is if the DNIS matches one of the first two exact numbers, it plays the background, sets the timeouts then goes on and plays thesound in the include and hangs up. What I want it to do is execute the stuff in the include ONLY if none of the exact matches ocurr. I would think this is the way it should work but I can't seem to make it happen. [incoming] Exten = 2145550001,1,Answer Exten = 2145550001,2,Wait(1) Exten = 2145550001,3,Background(MyGreeting) Exten = 2145550001,4,Timeout(30) Exten = 2145550001,5,DigitTimeout(3) Exten = 2145550002,1,Answer Exten = 2145550002,2,Wait(1) Exten = 2145550002,3,Background(MyGreeting) Exten = 2145550002,4,Timeout(30) Exten = 2145550002,5,DigitTimeout(3) Include = Pattern-Include [Pattern-Include] Exten = _8XXNXX,1,Answer Exten = _8XXNXX,2,Wait(1) Exten = _8XXNXX,3,Playback(NumNotConfigured) Exten = _8XXNXX,4,Hangup Private Label Wholesale Internet Access!http://www.YourOwnISP.com - Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; [EMAIL PROTECTED] Sent: Saturday, April 30, 2005 4:21 PM Subject: RE: [Asterisk-Users] Pattern Matching Like this: [dids] Exten = 2145550001,1,dial(SIP/6001) Exten = 2145550002,1,dial(SIP/6002) Exten = 2145550003,1,dial(SIP/6003) Include = default-did [default-did] Exten = _.,1,dial(SIP/6000) Seems pretty simple. I used this method of least/highest cost routing to choose my LD carrier. Should work the same though. http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo JojoSent: Saturday, April 30, 2005 3:08 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Pattern Matching Not sure what you mean exactly... Can you give me a hint? Private Label Wholesale Internet Access! http://www.YourOwnISP.com - Original Message - From: "Michael D Schelin" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 10:10 PM Subject: Re: [Asterisk-Users] Pattern Matching Hey Mojo, I'm thinking you might try using priorty 's to set some kind routing. just a thought.. Mojo Jojo wrote: We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in use/configured. I have been successful at setting up pattern matching as a catch all for 98 or so numbers not in use yet and I have been successful setting up the 2 numbers I want to make use of for now. Problem is, I can't use both at the same time! If I turn on the pattern matching then my greeting plays for the configured number, then the message plays for the invalid number (basically executing the extension with the pattern matching). I have read about sorting with pattern matching by using an include, I did this but it's not really helping. I have set a response timeout after the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.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 ___ 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 ___ As
[Asterisk-Users] TDM400P Power Connector
I have a TDM400P I am trying to install but I need a power connector extender to be able to get power into the card. In the meantime can the card run without the power connector if it has only one module on it? Thanks! Private Label Wholesale Internet Access! http://www.YourOwnISP.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] Pattern Matching
Not sure what you mean exactly... Can you give me a hint? Private Label Wholesale Internet Access! http://www.YourOwnISP.com - Original Message - From: Michael D Schelin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 10:10 PM Subject: Re: [Asterisk-Users] Pattern Matching Hey Mojo, I'm thinking you might try using priorty 's to set some kind routing. just a thought.. Mojo Jojo wrote: We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in use/configured. I have been successful at setting up pattern matching as a catch all for 98 or so numbers not in use yet and I have been successful setting up the 2 numbers I want to make use of for now. Problem is, I can't use both at the same time! If I turn on the pattern matching then my greeting plays for the configured number, then the message plays for the invalid number (basically executing the extension with the pattern matching). I have read about sorting with pattern matching by using an include, I did this but it's not really helping. I have set a response timeout after the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.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 ___ 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
[Asterisk-Users] Pattern Matching
We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in use/configured. I have been successful at setting up pattern matching as a catch all for 98 or so numbers not in use yet and I have been successful setting up the 2 numbers I want to make use of for now. Problem is, I can't use both at the same time! If I turn on the pattern matching then my greeting plays for the configured number, then the message plays for the invalid number (basically executing the extension with the pattern matching). I have read about sorting with pattern matching by using an include, I did this but it's not really helping. I have set a response timeout after the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.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] SIP registration behind Linksys WRT54G
I have a whole Asterisk server behind a wtr54gs. We have SPA-2000's registering from the Internet into it with no problems. Actually, we don't have it at the moment but did for several months. Not sure if this helps any or just adds to the confusion. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 10:24 PM Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G I've got a 7960 behind a Linksys wireless box and its working just fine with nat=yes in the sip.conf. Has been for over a year. Not sure of the model though. Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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
