Re: [Asterisk-Users] AGI Script: calleridnamelookup.agi
Has anyone goten this script to work with other reverse lookup providers. I also am sitting on a pri line that by standard only gives number returns. Thanks, On Thu, 19 Aug 2004 17:56:16 -0500, Greg Blakely <[EMAIL PROTECTED]> wrote: > Is anyone successfully using the AGI script calleridnamelookup.agi (or > anything similar) ? > > I get both name and number caller ID from my POTS line, but I'd save > money if I had them deliver ANI only. > > I've downloaded and installed the AGI script calleridnamelookup.agi, but > I always get > > -- Executing AGI("SIP/9525485560-5359", "calleridnamelookup.agi") in > new stack > -- Launched AGI Script > /var/lib/asterisk/agi-bin/calleridnamelookup.agi > -- AGI Script calleridnamelookup.agi completed, returning 0 > > I've even received that result calling in to my iconnect account, which > delivers only ANI information. > > I notice that the URL that it queries does not respond when I enter it > manually into a browser: > > http://www.anywho.com/qry/wp_rl/index.html?npa=719&telephone=471. A > box comes up that says "Fetching Results", and then the request times > out. > > Any idea how to structure the query on ANYWHO or how to use the script > with another reverse lookup service? > > Thanks in advance. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP-500 Network Problems
We currently use about 40 Polycom IP-500's and 35 Grandstream Bt101's at our company and have had the weirdest thing happen. When the power is unplugged to the supplied inline injector on the Polycom IP-500's ALL traffic on our local network just "dies" no packets are able to route any where. We are using 5 Netgear FSM726S (10/100 managed switch with 2 GBIC ports) Three of them are in a stack configuration linked over a short haul fiber link (~700 - 1000 ft) to 2 more switches which are in a stacked configuration. Any Idea's? Anyone heard of this problem before? Thanks, Matt Hohman New Heights Church 7913 ne 58th Ave Vancouver, WA 98665 360 750-7112 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AB1
Google? Hey thanks for the info I haven't seen that before. Wonders of modern technology. It's nice to use the list as a round table and get some insight. So they don't have disconnect detection I've heard of people using busy detection is this sufficient am I going to be wishing I paid the extra for the adit 600? Thanks for any help, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Sep 17, 2004, at 10:19 AM, Steven Critchfield wrote: On Fri, 2004-09-17 at 12:13, Matt Hohman wrote: We are looking at purchasing a carrier access ab1 channel bank forptsn interconnect. Has anyone used these before? Any insight? Have you heard about Google? If you put in some terms about what you want to know, and optionally give it a push in the right direction, it gives you all kinds of relevant information back. Technology is supposed to reduce our workload, not make it easier for you to zap the resolve of those who put forth effort. http://www.google.com/search?hl=en&ie=UTF-8&q=ab1+site%3Alists.digium.com 6 pages of results. Go read. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AB1
We are looking at purchasing a carrier access ab1 channel bank for ptsn interconnect. Has anyone used these before? Any insight? Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial-plan transfer
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? Any help would be great! Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote: We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything. Any idea's? Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan transfer. (h323 transfer)
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? Any help would be great! Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote: We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything. Any idea's? Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h.323 Transfer
We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything. Any idea's? Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] uniden Uip300 (UIP 200 STATUS)
Yes and no... We had to use the nat=never, We had soem issues with the phone freezing on call waiting this was fixed by the newest firmware 4.59a. But overall a good phone, we just need two lines. Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Sep 2, 2004, at 11:29 AM, Steve Maroney wrote: Sorry to be off topic of your post but did you have the problems documented on the link below with the UIP200 ? http://www.voip-info.org/wiki-UIP200 Thank you, Steve Maroney On Thu, 2 Sep 2004, Matt Hohman wrote: After having quite a bit of success with the UIP200 we decided to pick up a couple of UIP300 phones. After purchasing them we found that they were H.323 Ip phones. I searched on voip-info.org and have been unable to determine how exactly to setup this phone in asterisk. Any help would be graciously appreciated. Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uniden Uip300
After having quite a bit of success with the UIP200 we decided to pick up a couple of UIP300 phones. After purchasing them we found that they were H.323 Ip phones. I searched on voip-info.org and have been unable to determine how exactly to setup this phone in asterisk. Any help would be graciously appreciated. Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UIP300
After having a very good experiance with the UIP200 phones we decided to purchase a couple of UIP300 (2 line version) Only to releise that these phones are H.323 not sip. What is the difference between h.323 and sip. What needs to be done to get this phone working with asterisk? Thanks, Matt Hohman New Heights Church Vancouver, WA 360 694 4985 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ip10s Sip Firmware released
I have received a copy of the ip10s SIP firmware and have posted it on my site... www.mattsbooks.us/ip10s/ I have also posted the MGCP firmware for when you want to switch back... I have also put up a discussion board on the site please leave feedback with your experiences with the new firmware or ip10s issues Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]
[Asterisk-Users] Termination Provider
I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]
Re: [Asterisk-Users] Swissvoice ip10s
Yep I stayed and was able to get through to their ip-phone support in france. And with me only knowing english and the guy on the other end speaking "broken" english we kinda hashed out that it was a bad stick of flash ram in the phone. Communitech the USA provider for the phone is overnighting me a new one. AND emailing me the sip firmware for the mgcp phones. Thanks, Matt Hohman - Original Message - From: "Florian Overkamp" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 24, 2004 9:24 AM Subject: RE: [Asterisk-Users] Swissvoice ip10s > Hi, > > > -Original Message- > > Thanks! well after doing some other .cfg file changes I > > hardlocked the phone durring startup! Any ideas? (pushing > > 1,4,7 on powerup isn't helping) > > Ouch! Can you check if it is still fetching any config files from your > FTP-server at boot ? Might be your configs are corrupted somehow. If it is > not even doing that, you might just have to ship it back to SwissVoice and > have them fix it :-P > > Best regards, > Florian > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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] Swissvoice ip10s
Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Jun 24, 2004, at 12:33 AM, Florian Overkamp wrote: Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 "Transfer" NOINFO NOCONF TRUE NOSEQ set features new 2 "Operator" NOINFO NOCONF FALSE secretary> And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice ip10s
I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]