Re: [asterisk-users] Seemingly easy question: NPA/NXX
International numbers are variable length, so the timeout applies for those. North American National numbers are a fixed length. Generally, the phone company will collect 7, 10 or 11 digits for North American numbers. For example, I live in Minneapolis, MN. My number is 612-xxx-. I have free calling to 612, 651, 952, 763 and a few numbers in 507 and 320. If I dial 1, the phone company will collect 10 more digits. (The call may or may not go through if I dial 1+ a 10-digit local number depending on the carrier. MN regulations prohibit charging for local calls dialed as toll) If I dial 612, 651, 952, 763, 507 or 320, the phone company will wait for the remaining 7 digts as there are no numbers within area code 612 that start with those digits. Anything else will only be collected as 7 digts and assumed to be 612. Because of that, I can't dial a california number (for example), without dialing it as 1+. I wouldn't call it fancy, the phone company just knows what is a valid local number for you. Making a digit map in an ATA isn't that hard, you just need to think about what you want it to do. If you want to permit 10 digit dialing without the 1+ for long distance *and* support 7 digit local dialing, you'll need a timeout. There are also the N11 numbers, which of course should stop collecting after the second 1. --Shane Quoting Karl Fife [EMAIL PROTECTED]: Question: How does the local Telco know you're done dialing a seven digit number? Easy you may say: If your dial string begins with 1, the parser expects 11 digits total, otherwise seven, 011 is international. The reason suspect it's more complex is that: 1) International numbers can vary widely in length and 2) Our local analog Telco will route a ten digit NANP numbers with no leading 1 and with no terminator--seemingly instantly Obviously this could be done with 'timeouts'--implicitly 'sending' after a delay. But it works so well I suspect there's more logic in there. For example I have dozens of ATA's provisioned with timeouts, and I find it difficult or impossible to replicate the Telco dialing experience (Either the delay is too long, or you have frequent 'reorder' tones because it 'sent' before you were finished). Therefore I assume that there is something more 'fancy' going on. Can someone validate, debunk or clarify this? Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possibleto consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
I don't see where it is difficult to figure out. First of all, system keeps looking up on the table as user dial each number. When number starts with 1, expect USA. When number doesn't start with either 1 nor 0, expect USA too. When number starts with 011, and as country code and city code is identified, expect as many numbers as determined by country+city code (once you know country and city code, you know how many local digits to expect) CS Question: How does the local Telco know you're done dialing a seven digit number? Easy you may say: If your dial string begins with 1, the parser expects 11 digits total, otherwise seven, 011 is international. The reason suspect it's more complex is that: 1) International numbers can vary widely in length and 2) Our local analog Telco will route a ten digit NANP numbers with no leading 1 and with no terminator--seemingly instantly Obviously this could be done with 'timeouts'--implicitly 'sending' after a delay. But it works so well I suspect there's more logic in there. For example I have dozens of ATA's provisioned with timeouts, and I find it difficult or impossible to replicate the Telco dialing experience (Either the delay is too long, or you have frequent 'reorder' tones because it 'sent' before you were finished). Therefore I assume that there is something more 'fancy' going on. Can someone validate, debunk or clarify this? Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possibleto consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS Query Overload
Most distros come with a caching daemon. Still that's not really the point... If Asterisk has all of a sudden developed a habit of sending high volumes of nonsense DNS requests then it's a serious issue. Besides if the requests are different for each call the caching server is not going to help much and the downstream ISP is going to notice sooner or later. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi Stefan,
I have done what you told me to do, but nothing changed. Always the
same problem.
Here my extensions.conf
[originating_hind]
; Anonymous call
exten = _55tel_SIP.,1,Answer()
exten = _55tel_SIP.,2,SetCallerPres(prohib)
exten = _55tel_SIP.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _55tel_SIP.,4,Hangup()
[terminating_hind]
; Deny anonymous call
exten = _tel_SIP.,1,Goto(ivr_context,s,1)
[ivr_context]
exten = s,1,Answer()
exten = s,2,Set(caller={CALLERID(name)})
exten = s,n,GotoIf($["${caller}" = "Unknown"]?deny:allow)
exten = s,n(allow),Dial(SIP/${[EMAIL PROTECTED])
exten = s,n,Hangup()
exten = s,n(deny),Hangup()
Is there a problem in my file ?
Regards
Stefan Gofferje a crit:
[EMAIL PROTECTED] schrieb:
Thanks for help, but I don't understand what you say. How is it
possible to handle the error in the dialplan if my request return a 482
after entering Asterisk, but before accessing the dialplan ?
Ok :). I meant, you should handle the whole thing in the dialplan
without creating a loop.
A loop is when a request is originating from the same PBX as it is
directed to.
Example:
Your Asterisk is at IP 192.168.1.1.
You have a phone context and an IVR context
[phones]
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) ;Loop, VERY BAD!
exten = 4321,1,Goto(ivr_context,s,1) ;This is how it should be
[ivr_context]
exten = s,1,Background (welcome)
...
Terve,
Stefan
--
Rmi Druilhe
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi, [EMAIL PROTECTED] schrieb: I have done what you told me to do, but nothing changed. Always the same problem. If I understand your dialplan right, your * is still calling itself via SIP, right? This is what is called a loop. You should review your dialplan and replace all dial(SIP/[EMAIL PROTECTED]) by goto(respective_context,exten,pri). Or are you trying to call SIP clients which are registered to the box? In that case you don't dial(SIP/[EMAIL PROTECTED]) but dial(SIP/accountname) while accountname is what stands in [] for that client in your sip.conf. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon news online?
Hi, It's almost happening. Are there going to be any online feeds on Twitter, ScribbleLive or any audio or video streams? There are so many free tools to share your experiences in writing or via audio or video. Call a short report into Utterz.com. The #asterisk IRC channel, whatever. While I realize that when it's happening, you are involved, but how abput some reports from the room when you've had enough beer and didn't get lucky? Share your experiences as soon as you can, while they're fresh. Next Friday I hope to have a few returning Astricon people on the VUC to talk about the new Digium Beachball 2.0 or maybe even more significant experiences. The tweaker and the mousepads still rock. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon news online?
Hi, Voiceroute will be at Astricon and we will be twittering a lot on events at Astricon we plan to make small short videos on exhibits maybe tutorials happening during Astricon Keep updated with Astricon through Voiceroute Twitter http://www.twitter.com/voiceroute Voiceroute Youtube Channels http://youtube.com/user/voiceroute Ming On Mon, Sep 22, 2008 at 6:30 AM, randulo [EMAIL PROTECTED] wrote: Hi, It's almost happening. Are there going to be any online feeds on Twitter, ScribbleLive or any audio or video streams? There are so many free tools to share your experiences in writing or via audio or video. Call a short report into Utterz.com. The #asterisk IRC channel, whatever. While I realize that when it's happening, you are involved, but how abput some reports from the room when you've had enough beer and didn't get lucky? Share your experiences as soon as you can, while they're fresh. Next Friday I hope to have a few returning Astricon people on the VUC to talk about the new Digium Beachball 2.0 or maybe even more significant experiences. The tweaker and the mousepads still rock. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona http://www.astricon.net/2008/glendale/web/confTracks.php#t193 Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion, Javits Center, NYC http://druidweb20.eventbrite.com DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com UC 2.0 Video - Mozilla Ubiquity + Druid http://www.youtube.com/watch?v=f-5rDBPuGRc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon news online?
On Mon, Sep 22, 2008 at 12:30:52PM +0200, randulo wrote: Hi, It's almost happening. Are there going to be any online feeds on Twitter, ScribbleLive or any audio or video streams? There are so many free tools to share your experiences in writing or via audio or video. Call a short report into Utterz.com. The #asterisk IRC channel, #astricon seems rather empty right now. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
In fact, after entering in Asterisk for the first time, my call is redirected to an other component of my system. This other equiment redirect the same call to Asterisk a second time. It is something like this (it's an IMS architecture) : Softphone A -- Equiment -- Asterisk -- Equiment -- Asterisk -- Equipment -- Softphone B During my first passage in Asterisk, it sends an Invite with "Unknow" in field From. And for my second passage, it have to deny the call because of the sender identity which is "Unknow". Both of my softphones are not registered on Asterisk but on the equiment you can see before. Regards -- Rmi Druilhe Stefan Gofferje a crit: Hi, [EMAIL PROTECTED] schrieb: I have done what you told me to do, but nothing changed. Always the same problem. If I understand your dialplan right, your * is still calling itself via SIP, right? This is what is called a loop. You should review your dialplan and replace all dial(SIP/[EMAIL PROTECTED]) by goto(respective_context,exten,pri). Or are you trying to call SIP clients which are registered to the box? In that case you don't dial(SIP/[EMAIL PROTECTED]) but dial(SIP/accountname) while accountname is what stands in [] for that client in your sip.conf. Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setvar for outgoing SIP channels?
Hi! Using setvar in a peer configuration (sip.conf) I can set the channel variables for the incoming channel. Is there a similar method which allows me to load these variables also for outgoing channels (e.g. to load callee preferences)? thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem using AJAM on asterisk 1.4.17
Hello, I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not having much success. Right now the http server just listens on localhost:8088. I've used lynx and elinks for testing. I am able to get an Authentication accepted message using login, and I can view the stored auth cookie which is valid, but any attempt to use any other command immediate after results in an error response and Authentication Required message. I am not seeing any other errors in logs. My http.conf: [general] enabled=yes ;enablestatic=yes bindaddr=127.0.0.1 bindport=8088 And manager.conf: [general] enabled = yes webenabled = yes ;httptimeout = 10 port = 5038 ;bindaddr = 127.0.0.1 bindaddr = 0.0.0.0 displayconnects=off [testuser] secret = testpass read = all write = all deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 -- Thanks for any help that can be offered! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem using AJAM on asterisk 1.4.17
Hello, Try with : [testuser] secret = testpass read = all write = all deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Le lundi 22 septembre 2008 à 09:46 -0400, Jason Martin a écrit : Hello, I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not having much success. Right now the http server just listens on localhost:8088. I've used lynx and elinks for testing. I am able to get an Authentication accepted message using login, and I can view the stored auth cookie which is valid, but any attempt to use any other command immediate after results in an error response and Authentication Required message. I am not seeing any other errors in logs. My http.conf: [general] enabled=yes ;enablestatic=yes bindaddr=127.0.0.1 bindport=8088 And manager.conf: [general] enabled = yes webenabled = yes ;httptimeout = 10 port = 5038 ;bindaddr = 127.0.0.1 bindaddr = 0.0.0.0 displayconnects=off [testuser] secret = testpass read = all write = all deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 -- Thanks for any help that can be offered! -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 60 - Fax. : 01 41 38 99 70 http://www.proformatique.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?
may i noe wad can i do because my asterisk is working fine but the calls cannot proceed between 2 asterisk servers. hope anyone can help me solve this major problem. thanks a lot in advance Regards _ Get in touch with your inner athlete. Take the quiz. http://yourinnerathlete.windowslive.com?locale=en-sgocid=TXT_TAGLM_WLYIA_takequiz_sg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS Query Overload
What you should do, assuming that each DNS request is invalid and
returns nothing, is add a fake domain on your box that all of these
requests will point to. That is, if mydomain.com is the DNS name it's
looking up, add mydomain.com to a named server on the same box. Make
sure you include forwarders into the named.conf file pointing to your
DNS servers, as well as add a zone for mydomain.com.
zone mydomain.com {
type master;
file mydomain.com;
};
In the mydomain.com zone, add a wildcard A entry which will always
return the localhost address:
*.mydomain.com. IN A 127.0.0.1
Then point your DNS settings to localhost so all requests will go
through the local DNS server, and reset Asterisk.
In resolv.conf:
nameserver 127.0.0.1
search mydomain.com
This is essentially a hack...I've had the same issue in the past, but
was unable to get an answer as to why it's doing this.
Mik
Grey Man wrote:
Most distros come with a caching daemon.
Still that's not really the point... If Asterisk has all of a sudden
developed a habit of sending high volumes of nonsense DNS requests
then it's a serious issue. Besides if the requests are different for
each call the caching server is not going to help much and the
downstream ISP is going to notice sooner or later.
Regards,
Greyman.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi, [EMAIL PROTECTED] schrieb: In fact, after entering in Asterisk for the first time, my call is redirected to an other component of my system. This other equiment redirect the same call to Asterisk a second time. Hm, I suppose, your equipment is using reinvites for that redirection. The only idea to solve this I can think of would be having your equipment stay in the media path, i.e. making that redirection a brand new call. Then * shouldn't complain. But in my opinion that would be pretty ugly by means of scalability and ressources. Maybe a redesign of your callflow in general would be a better option. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
VoIP usage was legalized in India a few months back - http://www.voip-info.org/wiki/view/India+TRAI+Press+Release+legalizing+VOIP. Please let me know if I misunderstood it. BTW, I don't think I need a VoIP hardphone, the FXS slot would enable any ordinary phone to connect to Asterisk. Again, please let me know if I'm mistaken. Thanks. Best Regards, Hitesh On Sun, Sep 21, 2008 at 9:45 PM, Nhadie [EMAIL PROTECTED] wrote: i agree, you should check that, but i think if it's just for personal use they might allow it. regards, nhadie ram wrote: Hi Anotherthing you need to consider regulations in india Calling from Outside world to India PSTN Using and VOIP PBX is Illegal. Only allowed right now Calling From IP Phone to any part of the world see the regulation site dotindia.in http://dotindia.in for more information ram On Sat, Sep 6, 2008 at 9:26 PM, logan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello Everyone, Thanks for the answers, guys. DID won't work for me as I'm more interested in making calls using the line in India. Trixbox it's going to be my installation and I went through the recommended hardware on there site and I'm thinking of getting a Polycom 330. But can anyone tell me if there is a cheaper phone for me to use and which is well compatible with Trixbox (Polycom 330 is about $120 for me and something around $60-70 will be great)? I don't want all the fancy features, just something plain and simple to use. Coming to the FXO cards, I'm considering for Linksys SPA3102NA (successor of sipura 3000). I just want a second opinion from you guys if it's a good choice or there are better and cheaper options out there. Thanks a lot everyone. Best Regards, Hitesh - Original Message - From: Nhadie [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Saturday, September 20, 2008 1:09 AM Subject: Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line Hi Hitesh, Usually, subscribing to DID provider is a one way thing, they can call you to that number, but you cannot call out via that number. If you already have a pots line available, which means you are probably paying monthly for it already, might as well buy an fxo card and make use of the line. anyone in india can call you locally and you can call anyone in india using the same line, as everyone is suggesting, use trixbox, not much linux experience is required, just boot from the cd and let it install itself. you might need linux experience when you compile drivers for your fxo card though, but they usually come with instructions which is quite easy to follow. hth regards, nhadie logan wrote: Hi Jai, If I understand correctly then the DID will enable to call me on the hardphone connected to the Asterisk. Will it also enable me to call out using the PSTN line at my home in India from Canada? Thanks. Best REgards, Hitesh On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hitesh, If you dont have experience with Linux I would recommend you to use Trixbox, that will come with all the required packages and will do everythign for you. Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can buy DIDs that can come to your asterisk over the internet. Jai www.didforsale.com http://www.didforsale.com/ *Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com http://www.didforsale.com/ On Fri, Sep 19, 2008 at 9:18 AM, logan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello Ram, Thanks for the response. As I said there are too many options out there :). Could you help me in settling down on one? Something that will work with the phone lines in India is just fine for me. I don't have any or much Linux experience, but willing to play around, so any compatible distro will do for me. So once again: Which Linux distro is best with Asterisk? Which hardphone is the easiest to setup? Which fxo/fxs card I should go for? Thanks a lot guys. Best Regards, Hitesh On Thu, Sep 18, 2008 at 10:33 PM, ram [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Re: [asterisk-users] Seemingly easy question: NPA/NXX
When number starts with 011, and as country code and city code is identified, expect as many numbers as determined by country+city code (once you know country and city code, you know how many local digits to expect) ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. Luki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? If so than you must use time outs, if not than a dial plan can handle it. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIfTime and timezone specification
Hi! Is it possible to specify the timezone in the GotoIfTime application? E.g. I want to route the call if it is 9:00-10:00 in Austria/Vienna or 10:00 - 11:00 in New York. This is needed for example if the time based routing for the office in New York is done on an Asterisk server running another timezone. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
Ira schrieb: At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? If so than you must use time outs, if not than a dial plan can handle it. I think so. http://www.bundesnetzagentur.de/enid/7d0a4195f3524c5b04b7870e1f60635d,0/Ortsnetze/Struktur_und_Ausgestaltung_des_Nummernbereichs_fuer_Ortsnetzrufnummern_1gq.html http://www.bundesnetzagentur.de/enid/7d0a4195f3524c5b04b7870e1f60635d,0/Ortsnetze/Verzeichnisse_1gg.html But obviously this has nothing to do with the NANPA etc. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
Philipp Kempgen schrieb: Ira schrieb: At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? If so than you must use time outs, if not than a dial plan can handle it. I think so. http://www.bundesnetzagentur.de/enid/7d0a4195f3524c5b04b7870e1f60635d,0/Ortsnetze/Struktur_und_Ausgestaltung_des_Nummernbereichs_fuer_Ortsnetzrufnummern_1gq.html http://www.bundesnetzagentur.de/enid/7d0a4195f3524c5b04b7870e1f60635d,0/Ortsnetze/Verzeichnisse_1gg.html Strange session IDs. http://www.bundesnetzagentur.de/enid/Ortsnetze/Struktur_und_Ausgestaltung_des_Nummernbereichs_fuer_Ortsnetzrufnummern_1gq.html http://www.bundesnetzagentur.de/enid/Ortsnetze/Verzeichnisse_1gg.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
There are two bug reports with patches that might (?) be able to help: http://bugs.digium.com/view.php?id=7403 http://bugs.digium.com/view.php?id=12215 Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RNB (was: Re: Seemingly easy question: NPA/NXX)
Philipp Kempgen schrieb: Philipp Kempgen schrieb: Ira schrieb: At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? If so than you must use time outs, if not than a dial plan can handle it. http://www.bundesnetzagentur.de/enid/Ortsnetze/Struktur_und_Ausgestaltung_des_Nummernbereichs_fuer_Ortsnetzrufnummern_1gq.html http://www.bundesnetzagentur.de/enid/Ortsnetze/Verzeichnisse_1gg.html One of the documents says that if your number is 123456 (for example) you are allowed to use 123456x, 123456xx etc. OTOH it also says that the carriers are not obligated to make sure the additional digits make it to your line. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I can't call my remote users?
Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I am brand new to VOIP and even newer to Asterisk. I configured several PAP2T's and one SPA-962 to go home with our remote users. We had some trouble to begin with because at one of our locations they were hooking the SPA-962 into a WRT54G with a firewall so the phone was not getting an IP address. We ended up getting them to hook the phone into a switch that went to the MDF and the phone came online. The strange thing is that the phone would come online and then go offline: [Sep 12 22:11:04] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-1' is now Reachable. (117ms / 2000ms) -- Registered SIP '17114-3' at 74.46.72.86 port 60725 expires 3600 [Sep 12 22:11:04] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-2' is now Reachable. (126ms / 2000ms) -- Registered SIP '17114-4' at 74.46.72.86 port 60726 expires 3600 [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-3' is now Reachable. (109ms / 2000ms) -- Registered SIP '17114-5' at 74.46.72.86 port 60727 expires 3600 -- Registered SIP '17114-6' at 74.46.72.86 port 60728 expires 3600 [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-4' is now Reachable. (119ms / 2000ms) [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-5' is now Reachable. (108ms / 2000ms) [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-6' is now Reachable. (113ms / 2000ms) [Sep 12 22:12:08] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-1' is now UNREACHABLE! Last qualify: 117 [Sep 12 22:12:08] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-2' is now UNREACHABLE! Last qualify: 126 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-3' is now UNREACHABLE! Last qualify: 109 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-4' is now UNREACHABLE! Last qualify: 119 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-5' is now UNREACHABLE! Last qualify: 108 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-6' is now UNREACHABLE! Last qualify: 113 The thing that throws me off is then the peer became unreachable the user was still able to call; however, when I tried dialing the number I would get the out of service message. Any thoughts? Steve Anness ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I can't call my remote users?
Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I am brand new to VOIP and even newer to Asterisk. I configured several PAP2T's and one SPA-962 to go home with our remote users. We had some trouble to begin with because at one of our locations they were hooking the SPA-962 into a WRT54G with a firewall so the phone was not getting an IP address. We ended up getting them to hook the phone into a switch that went to the MDF and the phone came online. The strange thing is that the phone would come online and then go offline: [Sep 12 22:11:04] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-1' is now Reachable. (117ms / 2000ms) -- Registered SIP '17114-3' at 74.46.72.86 port 60725 expires 3600 [Sep 12 22:11:04] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-2' is now Reachable. (126ms / 2000ms) -- Registered SIP '17114-4' at 74.46.72.86 port 60726 expires 3600 [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-3' is now Reachable. (109ms / 2000ms) -- Registered SIP '17114-5' at 74.46.72.86 port 60727 expires 3600 -- Registered SIP '17114-6' at 74.46.72.86 port 60728 expires 3600 [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-4' is now Reachable. (119ms / 2000ms) [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-5' is now Reachable. (108ms / 2000ms) [Sep 12 22:11:05] NOTICE[3090]: chan_sip.c:12526 handle_response_peerpoke: Peer '17114-6' is now Reachable. (113ms / 2000ms) [Sep 12 22:12:08] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-1' is now UNREACHABLE! Last qualify: 117 [Sep 12 22:12:08] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-2' is now UNREACHABLE! Last qualify: 126 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-3' is now UNREACHABLE! Last qualify: 109 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-4' is now UNREACHABLE! Last qualify: 119 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-5' is now UNREACHABLE! Last qualify: 108 [Sep 12 22:12:09] NOTICE[3090]: chan_sip.c:15677 sip_poke_noanswer: Peer '17114-6' is now UNREACHABLE! Last qualify: 113 The thing that throws me off is then the peer became unreachable the user was still able to call; however, when I tried dialing the number I would get the out of service message. Any thoughts? Steve Anness ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
On Thu, 18 Sep 2008, Steve Anness wrote: So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. The text file that accompanies the voicemail (/var/spool/asterisk/voicemail/default//INBOX/msg.txt) contains a bunch of useful stuff: [message] origmailbox= context=macro-stdexten macrocontext=from-.net exten=s-NOANSWER priority=1 callerchan=SIP/sedwards-08d04618 callerid=xx xx origdate=Mon Sep 22 11:22:26 AM PDT 2008 origtime=1222107746 category= duration=6 Could you invoke a wrapper script using mailcmd (in voicemail.conf) to extract origmailbox and base your custom subject header based on that? I've never done this, but would like to know if it worked :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
(Replying to my own reply...)
On Mon, 22 Sep 2008, Steve Edwards wrote:
On Thu, 18 Sep 2008, Steve Anness wrote:
So here is the deal. I have an Asterisk server here at work that I
have recently taken over and the boss is wanting the server to do a
lot of things that it didn't do before. I have already configured much
of what he wanted including a voice messaging line where anyone can
call in and leave a message and then he would get that message in his
email. However, the boss wants his email subject to read something
like This is an urgent message through the HISG voice messaging
system so he knows that that message came through that number as
opposed to his voicemail box that already gets forwarded there. The
default is the [PBX]: New Message 10 in mailbox 0307. At second
glance he would know which voicemail box is his line but he wants
things to be different and so I am trying to make that happen.
I know there is the 'emailsubject' option. I haven't tried this yet
but my concern is that it will set the subject the same on every
single box (obviously what the command is designed for). I can I
customize a voicemail message so that if something comes in on our
0307 line it has a certain message and then we might get a message on
1942 line that we want a different subject.
The text file that accompanies the voicemail
(/var/spool/asterisk/voicemail/default//INBOX/msg.txt) contains a
bunch of useful stuff:
[message]
origmailbox=
context=macro-stdexten
macrocontext=from-.net
exten=s-NOANSWER
priority=1
callerchan=SIP/sedwards-08d04618
callerid=xx xx
origdate=Mon Sep 22 11:22:26 AM PDT 2008
origtime=1222107746
category=
duration=6
Could you invoke a wrapper script using mailcmd (in voicemail.conf) to
extract origmailbox and base your custom subject header based on that?
I've never done this, but would like to know if it worked :)
The msg000.txt file is not available to the command invoked by mailcmd
so scratch that suggestion.
The mailcmd variable in voicemail.conf appears to be a per user
variable (1.2). It's stored in the ast_vm_user linked list.
The emailsubject variable is global, but it is munged using
prep_email_sub_vars() and pbx_substitute_variables_helper(). You may
have some luck setting emailsubject to something like [PBX]: ${URGENT}
New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} and the setting URGENT
appropriately in the dialplan. I didn't try it, but it may work.
Failing the above, the mailcmd variable invokes the wrapper with the
SMTP headers, the body, and (if any) attachment (base64 encoded) all as a
stream on stdin. All you need to do is read stdin and mung the Subject
header as you feed everything off to sendmail -t.
Failing either of the above approaches, you can always break out your
code hatchet :)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem using AJAM on asterisk 1.4.17
Thanks for the reply. I tried the option below but it did not yield any different results. --- Hello, Try with : [testuser] secret = testpass read = all write = all deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Le lundi 22 septembre 2008 ? 09:46 -0400, Jason Martin a ?crit : Hello, I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not having much success. Right now the http server just listens on localhost:8088. I've used lynx and elinks for testing. I am able to get an Authentication accepted message using login, and I can view the stored auth cookie which is valid, but any attempt to use any other command immediate after results in an error response and Authentication Required message. I am not seeing any other errors in logs. My http.conf: [general] enabled=yes ;enablestatic=yes bindaddr=127.0.0.1 bindport=8088 And manager.conf: [general] enabled = yes webenabled = yes ;httptimeout = 10 port = 5038 ;bindaddr = 127.0.0.1 bindaddr = 0.0.0.0 displayconnects=off [testuser] secret = testpass read = all write = all deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 -- Thanks for any help that can be offered! -- ?Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 60 - Fax. : 01 41 38 99 70 http://www.proformatique.com -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EM wink/no audio
I am preparing to connect an asterisk box with a redfone fonebridge to a T1 service provider. I am doing this by testing first with another asterisk and a Sangoma card playing the role of telco. I formerly had this test configuration operating flawlessly as a PRI connection. But I discovered that I will need to use EM, thus I've chosen the parameters as described in the subject line. So far, I am able to initiate a call from the Sangoma/telco side to the fonebridge side, and basic robbed bit/ABCD call supervision seems to work. That is, I can see the flags going up and down on but sides with zttool, and a Zap channel is allocated on each side. But it seems that DNIS is not going through, and so I had to fudge it by creating a s extension on the called side to pass the call through to a SIP telephone. When the call is answered, the caller hears silence, and the call recipient hears a soft squeal. Am I being reasonable in assuming that the absence of a valid audio stream is the likely reason that the called number is not being passed through successfully? In any case, what type of stuff should I be looking for to diagnose this? FYI, the calling side issues a log message that seems relevant, but it's precise implications elude me: chan_zap.c: Ignoring wink on channel 1 (see below). I hope someone can give clue on these matters. [Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/366-a41a4560, ZAP/g1/1222333|300|wW) in new stack [Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Dialing '1222333' [Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Deferring dialing... [Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Called g1/1222333 [Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Ignoring wink on channel 1 [Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Sent deferred digit string: T1222333w [Sep 22 11:27:19] VERBOSE[27487] logger.c: -- Hungup 'Zap/1-1' smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem using AJAM on asterisk 1.4.17
Le lundi 22 septembre 2008 à 16:10 -0400, Jason Martin a écrit : Thanks for the reply. I tried the option below but it did not yield any different results. Have you read this ? http://www.the-asterisk-book.com/unstable/manager-interface-ajam.html Have you reading the log on your CLI when you launch your lynx ? What is the url for lynx ? Sylvain -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 64 - Fax. : 01 41 38 99 70 Email : [EMAIL PROTECTED] - http://proformatique.com/ signature.asc Description: Ceci est une partie de message numériquement signée ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ast_func_write: Function not registered
hi all , please need help for an
asterisk version 1.4.21.2
i created a write func odbc list records files in sql table:
[R]
dsn=connector
write=INSERT INTO ast_records (filename,caller,callee,dtime) VALUES
('${ARG1}','${ARG2}','${ARG3}','${ARG4}')
prefix=M
and set it in dialplan :
exten = _0X.,n,Set(
M_R(${MIXMONITOR_FILENAME}\,${CUSER}\,${EXTEN}\,${DTIME})= )
i tried to add preload = func_odbc.so to load before extentions.conf
but when i excute it still have ast_func_write: M_R Function not registered:
Executing [EMAIL PROTECTED]:6] Set(IAX2/555-10991,
M_R(/records/23-09-08-02-22_555_0343434.gsm,555,0343434,23-09-08-02-22)= )
in new stack
[Sep 23 02:22:14] ERROR[3261]: pbx.c:1564 astr _func_write: Function M_R
not registered
any suggestions ??
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone
Hi,
On my call back system, I have the script as follows:
[calback]
exten = s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten = s,n,Set(CALL=${CALLERID(number)})
exten = s,n,Set(DESTINATION=myCallback.2000.1)
exten = s,n,Set(SLEEP=5)
exten = s,n,System(/var/lib/asterisk/bin/callback ${CALL} ${DESTINATION}
${SLEEP} )
exten = s,n,Hangup
The idea behind this system is that the script picks up the call, notes down
the caller's number, and hangs it immediately. Then the caller gets a call
back.
But what is happening is that cell phone callers are still being charged for
calling into this callback context.
How can I avoid this? I want cell phone users to not get charged for the
call back.
--
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Send us your suggestions on exhibits tutorials to cover (video) at Voiceroute
Hi all, We are compiling a list of exhibits tutorials to cover at Voiceroute. We will be twittering doing impromptu videos. We are looking for votes suggestions on what people would like us to cover. Pls send suggestions to [EMAIL PROTECTED] The team at Voiceroute have our blackberries iphones + audio recorders + digital video camera ready to cover the action at Astricon Pictures available at http://picasaweb.google.com/mgyong/ Video at http://youtube.com/voiceroute Twitter at http://twitter.com/voiceroute Ming -- Forwarded message -- From: Ming Yong [EMAIL PROTECTED] Date: Mon, Sep 22, 2008 at 6:58 AM Subject: Re: [asterisk-users] Astricon news online? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi, Voiceroute will be at Astricon and we will be twittering a lot on events at Astricon we plan to make small short videos on exhibits maybe tutorials happening during Astricon Keep updated with Astricon through Voiceroute Twitter http://www.twitter.com/voiceroute Voiceroute Youtube Channels http://youtube.com/user/voiceroute Ming On Mon, Sep 22, 2008 at 6:30 AM, randulo [EMAIL PROTECTED] wrote: Hi, It's almost happening. Are there going to be any online feeds on Twitter, ScribbleLive or any audio or video streams? There are so many free tools to share your experiences in writing or via audio or video. Call a short report into Utterz.com. The #asterisk IRC channel, whatever. While I realize that when it's happening, you are involved, but how abput some reports from the room when you've had enough beer and didn't get lucky? Share your experiences as soon as you can, while they're fresh. Next Friday I hope to have a few returning Astricon people on the VUC to talk about the new Digium Beachball 2.0 or maybe even more significant experiences. The tweaker and the mousepads still rock. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona http://www.astricon.net/2008/glendale/web/confTracks.php#t193 Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion, Javits Center, NYC http://druidweb20.eventbrite.com DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com UC 2.0 Video - Mozilla Ubiquity + Druid http://www.youtube.com/watch?v=f-5rDBPuGRc -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona http://www.astricon.net/2008/glendale/web/confTracks.php#t193 Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion, Javits Center, NYC http://druidweb20.eventbrite.com DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com UC 2.0 Video - Mozilla Ubiquity + Druid http://www.youtube.com/watch?v=f-5rDBPuGRc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN Simulator
Hi, I have Asterisk setup to run on SS7, and I would like to test it out before getting the line from my telco. Is there any testing or simulation tool that I can buy to simulate a E1/SS7 link? Could anyone give some suggestions? Thanks alot for your help in advance. Regards, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN Simulator
Another asterisk box set up to be the network side of that link? mark morreny wrote: Hi, I have Asterisk setup to run on SS7, and I would like to test it out before getting the line from my telco. Is there any testing or simulation tool that I can buy to simulate a E1/SS7 link? Could anyone give some suggestions? Thanks alot for your help in advance. Regards, Mark ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
