Re: [asterisk-users] call to mobiles and it is turn off
simple and perfect now is works fine! thanks Eric ManxPower Wieling wrote: Remove the r option to Dial. Bruno Castelo Branco wrote: Hi all When I call to any mobile and the device is power off the asterisk keep ringing and I not able to hear the tradicional message saying this mobile is power off. When I call from a normal analogic line I got the message. Somebody have some suggestion to enable asterisk to identify turn off devices and pass the message to peer? otherwise when somebody call to some mobile always think is ringing and not power off. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] say I wish to run tail command on messages file to pick up if any channels unavailable messages appear.
Can I use grep ? Tried but not working. please help Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Problem chan_sip.c: Call''from''to extension rejected because extension not found.
Hi All, how are you? I would like to know from you if the problem can be below is a BUG of the asterisk-1.4.21. I did an upgrade version of asterisk-1.2.18 for the version of asterisk-1.4.21 and now, when users try to sip friend outgoing calls through Polycom IP 330 appliances can not be the traditional way or with the telephone handset in his hand and digit dialing digit to receive the following information in the CLI asterisk: chan_sip.c: Call''from''to extension rejected because extension not found. Example: [Dec 12 10:05:44] NOTICE [23,330] chan_sip.c: Call from'19703 'extension to'00' rejected because extension not found. [Dec 12 10:05:47] NOTICE [23,330] chan_sip.c: Call from'19703 'extension to'00' rejected because extension not found. [Dec 12 10:05:49] NOTICE [23,330] chan_sip.c: Call from'19703 'extension to'00' rejected because extension not found. And sending all digits at once the connection is completed successfully. The following is the configuration of sip friend: [19703] type = friend context = DLD Secret = X host = dynamic dtmfmode = rfc2833 regexten = 19703 CallerId = 19703 19703 callgroup = 25 pickupgroup = 25 nat = yes amaflags = billing Could you help me? Best Regards Josue ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FBI issues VoIP security warning on Asterisk --which version?
Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send a call to a Polycom SIP phone with NOcallerid whatsoever
Thanks, that makes plenty of sense. I thought I could only check if a phone as busted it's call-limit, but I just tested and it works well. Thank you! Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexander Lopez Sent: Friday, December 12, 2008 1:49 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to send a call to a Polycom SIP phone with NOcallerid whatsoever If the page was answered on the Polycom then it would NOT show up as a missed call, a received call yes but not a missed call. If you are getting missed calls from the page application, the users are probably ON the phone when you page, if so you should put something in your dialplan that checks to see if the phone is in use and if so do not send a page call to the phone. Alex _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 7:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to send a call to a Polycom SIP phone with NOcallerid whatsoever I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a missed call shown on the screen. I have access to the Polycom phone.cfg file, and obviously to the Asterisk .conf files. Anything I can do? Can I send a SIP header to say don`t show any call data on the screen? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Are Cisco SIP phones still non-localizable with an Asterisk server ?
Hi, I heard some time ago that, when running a SIP firmware, Cisco hardphones needed a Cisco call manager to get localized (ie non-english) menus ? Is it still true ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FBI issues VoIP security warning on Asterisk --which version?
On 14:51, Fri 12 Dec 08, Khaled Chehab wrote: Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards Hi, See this listpost: http://lists.digium.com/pipermail/asterisk-users/2008-December/223172.html -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with Asterisk on Ubuntu
Scott Berry schrieb: I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run asterisk -r at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It sure does exist. I also see I am running like a hundred different modules according to /var/log/asterisk/messages. Simply re-posting a question does not help. http://lists.digium.com/pipermail/asterisk-users/2008-December/223239.html Did you already try any of the suggestions? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] say I wish to run tail command on messages file to pick up if any channels unavailable messages appear.
Shaun Wingrin schrieb: Can I use grep ? Tried but not working. tail -f /var/log/asterisk/messages | grep 'channels unavailable' 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] say I wish to run tail command on messages file to pick up if any channels unavailable messages appear.
Try this first: Cat /var/log/asterisk/messages|grep channels unavailable Once you get grep output from this, changing the grep on the tail command should produce the desired results. Since the tail -f is a dynamic situation, it is much easier to make the test on the fixed cat command. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, December 12, 2008 7:30 AM To: Asterisk Users Subject: Re: [asterisk-users] say I wish to run tail command on messages file to pick up if any channels unavailable messages appear. Shaun Wingrin schrieb: Can I use grep ? Tried but not working. tail -f /var/log/asterisk/messages | grep 'channels unavailable' 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FBI issues VoIP security warning on Asterisk--which version?
This seemed to be specific to 1.4.19 and prior. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled Chehab Sent: Friday, December 12, 2008 6:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] FBI issues VoIP security warning on Asterisk--which version? Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] prepaid solution
Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context. [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work.until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FBI issues VoIP security warning on Asterisk --which version?
On Dec 12, 2008, at 8:16 AM, Michiel van Baak wrote: On 14:51, Fri 12 Dec 08, Khaled Chehab wrote: Dear All FBI issues VoIP security warning on Asterisk -- but which version? Any one know which version ? Regards Hi, See this listpost: http://lists.digium.com/pipermail/asterisk-users/2008-December/223172.html -- Michiel van Baak And for a rant, see this: http://www.voiptechchat.com/voip/146/fbi-security-warnings-and-voip/ -Fred Posner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get first month of trixbox free
Hi Don, Don Kelly wrote: Caution---top posting. It works for me--ignore it if you like. Without forgiving Michael's commercial message to *-users, perhaps we can punish your top posting by highlighting your gross insensitivity to the physically challenged! :-) And (flame follows) we would do it using careful use of English grammar and spelling, especially if we were The only major error I found were threw instead of through. Perhaps that is forgivable for someone using a handle of creepyBLINDy? N'est pas? regards, Drew * From: * asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Babcock, Michael Alex* * hi; threw the end of the year we are running a promo, when ordering any package on http://gwhosting.net including our vps servers and trixbox servers, you can get your first month off. Yes, that's right, enter 30free with out the quote signs into the coupon code field during checkout to get your first month free. Give us a try, you won't be sorry. Your security is our number one priority. GW Hosting, your dedicated home on the web: http://gwhosting.net 30free does truly get you your free month. Stop at any time during your first month and you won't be charged any more, no strings attached. Well, wait there is one string, you have to go to http://gwhosting.net and sign up using 30free to get the free month. thanks Michael Babcock Michael Babcock Owner of GW Hosting, http://www.gwhosting.net http://www.gwhosting.net/ For information on what I may be doing at the moment, please feel free to visit my blog, twitter or brightkite at the following links: Twitter: http://www.twitter.com/creepyblindy Blog: http://www.gwfans.net http://www.gwfans.net/ Brightkite: http://brightkite.com/people/creepyblindy -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ignoring context= in sip.conf
On Fri, Dec 12, 2008 at 08:11:51PM +1300, Michael wrote: I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Could you please post either your sip.conf file or a minimal variation of it that has this issue? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ignoring context= in sip.conf
Did you make a [xyz] context in extensions.conf? if the sip.conf doesn't find the content, it drops back to default. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, December 12, 2008 11:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk ignoring context= in sip.conf On Fri, Dec 12, 2008 at 08:11:51PM +1300, Michael wrote: I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Could you please post either your sip.conf file or a minimal variation of it that has this issue? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] docs for rxfax in 1.4 or app_fax in 1.6?
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
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[asterisk-users] multivoip bogen mp130
has anyone used a multivoip device from bogen MP130? I am looking for information on how to configure it for use with asterisk. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe echo problems with more than two participants
Alessandro Russo wrote: we are using Asterisk 1.4.18.1 http://1.4.18.1/ on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? Alessandro, Are you certain that the echo isn't being introduced by someone on the conference using a speakerphone? This would cause what is known as acoustic echo http://en.wikipedia.org/wiki/Echo_cancellation#Acoustic_echo and it's always my first suspect in a situation like the one you are describing. This is not a problem that is specific to Asterisk and I'm fairly certain there is nothing that can be done within your configuration to correct it. Instructing the conference participants to mute their phones when they aren't speaking or to use their handsets should reduce acoustic echo. Some phones http://www.voip-info.org/wiki/view/Uni-Ta+Technology also claim to have a full-duplex speakerphone with advanced acoustic echo cancellation, but caveat emptor. That said, I'm not an expert on echo cancellation and I have an installation where the users are making similar complaints about echo during conference calls. I'd greatly appreciate it if anyone on the list corrected any misunderstandings that I might have on the subject. As an aside, how is the timing on your conference server. The MeetMe application relies on it to mix the audio in conferences. You should get at least 99.98% output from zttest (as shown below) or the audio quality will suffer. This is an overall quality issue and is not necessarily related to your echo problems. [r...@astconf ~]# zttest Opened pseudo zap interface, measuring accuracy... 99.999413% 99.995407% 99.995499% 99.998047% 99.996483% 99.997849% 99.999008% ... --- Results after 107 passes --- Best: 100.000 -- Worst: 99.995 -- Average: 99.997687, Difference: 99.997815 Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 9:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] prepaid solution
prepaid solution for what? 2008/12/12 BERGANZ François franc...@acropolistelecom.net Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
After some research, it seems that asterisk builds a dynamic context called [park-dial] and puts a callback for the parker into line 1, so this would have to be a patch/workaround. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 12, 2008 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 9:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context. [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work.until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MSet()
How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSet()
Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? Philipp Kempgen It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSet()
Mark Michelson wrote: Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? Philipp Kempgen It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. Mark Michelson An even better answer is in the UPGRADE-1.6.txt document in the Asterisk source: * The behavior of the Set application now depends upon a compatibility option, set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To use the new behavior, which permits variables to be set with embedded commas, set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both behaviors at the same time, if you switch to using MSet if you want the old behavior. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSet()
Mark Michelson schrieb: Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. I was confused because even in 1.4 The use of Set to set multiple variables at once and the g flag have both been deprecated. Please use multiple Set calls and the GLOBAL() dialplan function instead. There's nothing wrong with Set(foo=hello); Set(bar=world); so I was just wondering why 1.6 comes with a Multi-Set() application. Never mind. Thanks. 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ring back tone
Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSet()
On Friday 12 December 2008 16:05:38 Philipp Kempgen wrote: Mark Michelson schrieb: Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. I was confused because even in 1.4 The use of Set to set multiple variables at once and the g flag have both been deprecated. Please use multiple Set calls and the GLOBAL() dialplan function instead. There's nothing wrong with Set(foo=hello); Set(bar=world); so I was just wondering why 1.6 comes with a Multi-Set() application. Because we were roundly criticized for removing what some considered to be critical functionality. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
No, not on FXO ports. On FXO ports Asterisk considers the call answered as soon as dialing is finished. Asterisk has no way to detect when the far end answers when using FXO ports. michel freiha wrote: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] docs for rxfax in 1.4 or app_fax in 1.6?
On Friday 12 December 2008 12:08:55 sean darcy wrote:
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
i'm working on a email - fax gateway right now to do just that. it works
well, but is unpolished and basically undocumented at this point. you can see
the work in svn here: http://messinet.com/viewvc/asterisk-fax-gw/trunk/
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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Re: [asterisk-users] MSet()
On Friday 12 December 2008 15:41:54 Mark Michelson wrote: would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. what about Set(ARRAY(var1,var2)=value1,value2) ? is the MSet() app a better/quicker way to do this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
Philipp Kempgen wrote: michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? Much like violence and herding of llamas, the r option to Dial (and the Ringing app) almost never solve the problem they are intended to solve and frequently cause more, usually unforeseen, problems. Just say No! to r. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Foundry serverIron does support SIP and its ASIC not a linux box Load balancer like F5, Refer to Chapter 10 (page 677) of ServerIron manual. It explains everything in detail. Also you may need to play with source nat a little bit to make your specific configuration work, but it should work, at least in theory. On Thu, Nov 20, 2008 at 10:25 AM, Alex Balashov abalas...@evaristesys.comwrote: SIP wrote: As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. I assume that is because there is no way RFC-supported way to insert a cookie into a SIP session that persists throughout the entire exchange with a client, including all in-dialog requests, subsequent sessions, etc? The only way I know of to make a cookie stick on the UAC side is to put an LR parameter into the route set, but that will only last within a dialog. So, I'm assuming certain SIP proxies had proprietary ways of getting around that in order to work with F5? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
At this point, it's six one, half dozen the other, although that may change with time. Kamailio and SER appear to be joining forces. But it's mostly a matter of your affinity with the community and the various political forces and personalities at this point. I personally am sticking with the Kamailio camp because I think they are doing a better job of creating a stable business environment around the project and doing things that are important to big-name adopters who are far more concerned about having something they can lean on than about coding, coding and coding. On the other hand, the OpenSIPS camp has proposed some very radical and potentially beneficial architectural changes if they are actually carried through. Still, at this point in time, six one half-dozen the other, especially if you're talking about the core and stateful (tm module) functionality needed for things that fall under the rubric of front-ending Asterisk. Steve Edwards wrote: One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
But, just to be clear, the answer to your question - or more precisely, the answer to the question underlying your question (WHY are you using what you're using?) - is fundamentally political in its essence. Most aspects of the case for one or the other have little to do with technology. At this time. Alex Balashov wrote: At this point, it's six one, half dozen the other, although that may change with time. Kamailio and SER appear to be joining forces. But it's mostly a matter of your affinity with the community and the various political forces and personalities at this point. I personally am sticking with the Kamailio camp because I think they are doing a better job of creating a stable business environment around the project and doing things that are important to big-name adopters who are far more concerned about having something they can lean on than about coding, coding and coding. On the other hand, the OpenSIPS camp has proposed some very radical and potentially beneficial architectural changes if they are actually carried through. Still, at this point in time, six one half-dozen the other, especially if you're talking about the core and stateful (tm module) functionality needed for things that fall under the rubric of front-ending Asterisk. Steve Edwards wrote: One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
Have you checked out OpenSBC (www.voip-info.org/wiki/view/*OpenSBC)?* On Fri, Dec 12, 2008 at 6:19 PM, Steve Edwards asterisk@sedwards.comwrote: One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
Also, both projects are open source, so they can (and do) take patches from each other both for bug fixes and for new features. The smart project will take the good stuff from the other while simultaneously doing a better job of providing a commercial and political ecosystem that leads to serious adoption and the creation of new value. From my point of view, some of the most innovative contributors of code are on the OpenSIPS side (mostly the folks at the disposal of Bogdan-Andrei Iancu/voice-system.ro), but most of the judicious and sophisticated project management talent from the OpenSER group stayed with Kamailio, including Daniel-Constantin Mierla, Elena-Ramona Modroiu, Henning Westerholt, and Juha Heinanen - some of the other great minds that are strategically essential. But arguing about which group is a better group of people to work with is just going to provoke a needless ad hominem flame war and further bitterness and strife. Alex Balashov wrote: But, just to be clear, the answer to your question - or more precisely, the answer to the question underlying your question (WHY are you using what you're using?) - is fundamentally political in its essence. Most aspects of the case for one or the other have little to do with technology. At this time. Alex Balashov wrote: At this point, it's six one, half dozen the other, although that may change with time. Kamailio and SER appear to be joining forces. But it's mostly a matter of your affinity with the community and the various political forces and personalities at this point. I personally am sticking with the Kamailio camp because I think they are doing a better job of creating a stable business environment around the project and doing things that are important to big-name adopters who are far more concerned about having something they can lean on than about coding, coding and coding. On the other hand, the OpenSIPS camp has proposed some very radical and potentially beneficial architectural changes if they are actually carried through. Still, at this point in time, six one half-dozen the other, especially if you're talking about the core and stateful (tm module) functionality needed for things that fall under the rubric of front-ending Asterisk. Steve Edwards wrote: One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Origination with RDNIS
I am looking for a VOIP provider that can offer origination and provide the RDNIS with each call. I am not looking for any large volume commitment. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Country numbering plan resources
Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21 and 29 followed by 6, 7 or 8 digits - Vodafone GSM area code 27 followed by 6 or 7 digits - NZ Telecom CDMA note that there is number portability so the above is a guide. NZ Landline: area code 3, 4, 6, 7 and 9 followed by 7 digits (first digit will be in the range of 2-9) NZ toll free: area code 508 and 800 followed by 6 digits NZ premium: area code 900 - though I doubt any of you will be routing these calls AU cellular: area code 4 followed by a 2 digit network code, and then a 6 digit number Networks include: Optus, Telstra, 3, Vodafone, Virgin and others. All use GSM and there is number portability. AU landline: area code 2, 3, 7 and 8 followed by 8 digits (first digit will be in the range of 2-9) AU toll free: area code 1300 or 1800 followed by 6 digits OR area code 13 followed by 4 digits. AU premium: I'm not sure though someone present may fill us in. Following is the list of countries I need information on: ; ANDORRA ; ARGENTINA ; AUSTRIA ; BAHAMAS ; BELGIUM ; BRAZIL ; BULGARIA ; CANADA ; CHILE ; CHINA ; COLOMBIA ; CROATIA ; CYPRUS SOUTH ; CZECH REPUBLIC ; DENMARK ; ESTONIA ; FRANCE ; GERMANY ; GREECE ; GUADELOUPE ; GUAM ; HONG KONG ; HUNGARY ; ICELAND ; INDONESIA ; IRELAND ; ISRAEL ; ITALY ; JAPAN ; JORDAN ; SOUTH KOREA ; LUXEMBOURG ; MALAYSIA ; MARIANA ISLANDS ; MEXICO ; MONACO ; NETHERLANDS ; NORWAY ; PANAMA ; PERU ; PERU LIMA ; POLAND ; PORTUGAL ; PUERTO RICO ; ROMANIA ; RUSSIA ; SAN MARINO ; SINGAPORE ; SLOVAKIA ; SLOVENIA ; SPAIN ; SWEDEN ; SWITZERLAND ; TAIWAN ; THAILAND ; TURKEY ; UNITED KINGDOM ; UNITED STATES ; VENEZUELA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
One of the problems you'll run into is that in larger countries like the US, and/or countries with greater amounts of telecom interconnection, competition and deregulation, this information cannot be reduced simply to a convenient algorithm. The North American Numbering Plan (www.nanpa.com) does provide some basic standards for valid numbers, but aside from that, there exists no special numerological distinction between incumbent and competitive, fixed-line and mobile, or VoIP, and extensive number portability throws even more complexity into the mix. I'm not saying it can't be done - just be aware that the undertaking you're proposing is very complicated, and the information would come from innumerable data sources (a great deal of them commercial and expensive) and a bewilderingly overlapping array of standards bodies. For instance, something like this: NZ Cellular: area code 21 and 29 followed by 6, 7 or 8 digits - Vodafone GSM area code 27 followed by 6 or 7 digits - NZ Telecom CDMA note that there is number portability so the above is a guide. ... sounds like a laughably, impossibly simplistic formula to a North American reader. And I can't imagine the situation in many other countries is much simpler. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
If you want an idea of what the setup looks like in NANPA land, enjoy this convenient spreadsheet: http://www.nanpa.com/nanp1/allutlzd.zip Notice that there is no discernable pattern to the number space allocated to a particular flavour of carrier. And do beware that these are 10,000 code blocks only; in pooled areas (most metropolitan areas, and an increasingly large number of areas) these get split up into 1,000-code blocks and the information from that comes from National Pooling/Neustar (www.nationalpooling.com). Consequently, routing or analysing anything by 10,000 blocks is becoming an increasingly useless practise. Oh, and don't forget the byzantine properties of portability here. Alex Balashov wrote: One of the problems you'll run into is that in larger countries like the US, and/or countries with greater amounts of telecom interconnection, competition and deregulation, this information cannot be reduced simply to a convenient algorithm. The North American Numbering Plan (www.nanpa.com) does provide some basic standards for valid numbers, but aside from that, there exists no special numerological distinction between incumbent and competitive, fixed-line and mobile, or VoIP, and extensive number portability throws even more complexity into the mix. I'm not saying it can't be done - just be aware that the undertaking you're proposing is very complicated, and the information would come from innumerable data sources (a great deal of them commercial and expensive) and a bewilderingly overlapping array of standards bodies. For instance, something like this: NZ Cellular: area code 21 and 29 followed by 6, 7 or 8 digits - Vodafone GSM area code 27 followed by 6 or 7 digits - NZ Telecom CDMA note that there is number portability so the above is a guide. ... sounds like a laughably, impossibly simplistic formula to a North American reader. And I can't imagine the situation in many other countries is much simpler. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008 16:24:56 you wrote: One of the problems you'll run into is that in larger countries like the US, and/or countries with greater amounts of telecom interconnection, competition and deregulation, this information cannot be reduced simply to a convenient algorithm. The North American Numbering Plan (www.nanpa.com) does provide some basic standards for valid numbers, but aside from that, there exists no special numerological distinction between incumbent and competitive, fixed-line and mobile, or VoIP, and extensive number portability throws even more complexity into the mix. I'm not saying it can't be done - just be aware that the undertaking you're proposing is very complicated, and the information would come from innumerable data sources (a great deal of them commercial and expensive) and a bewilderingly overlapping array of standards bodies. Yes, but calls to the USA and Canada landline/cellular cost the same. I need as many countries in the list that I can get info on because in many cases cellular calls and landline calls are priced differently and I need to make routing distinctions in my dial plan. Yes you are correct, Australia and New Zealand are an easy plan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: Yes, but calls to the USA and Canada landline/cellular cost the same. What? No, they don't. They absolutely do not; there are many, many different intercarrier compensation tiers here depending on operating area and the applicable tariffs arising from regulatory classification. Termination to mobile carriers most definitely does not cost the same as fixed-line, and termination to fixed-line varies widely depending on interconnection agreements and who is doing the transport and the switching where. *You* may be getting a blended rate plan that amortises all the damage into one convenient rate, but rest assured that it doesn't cost the underlying carrier the same rate all around. If you get a plan from someone that gives you a rate deck by LATA, Tier or OCN, you will find that to be true. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008, Michael wrote: I'm not saying it can't be done - just be aware that the undertaking you're proposing is very complicated, and the information would come from innumerable data sources (a great deal of them commercial and expensive) and a bewilderingly overlapping array of standards bodies. Yes, but calls to the USA and Canada landline/cellular cost the same. I need as many countries in the list that I can get info on because in many cases cellular calls and landline calls are priced differently and I need to make routing distinctions in my dial plan. In general you don't need to worry about that, as when you go to buy your routes, the splits are given to you. For example, though you have split up New Zealand nicely I don't need that information, as the termination provider I buy New Zealand from gives me one price for what they deem proper (01164) and another several for what they deem mobile (01164900, 011648, 011642). Whatever destination is dialed simply picks the route that it most matches, and I know what the charges are. This does mean that you have to stay in synch with any changes your upstream termination providers make to their dial plans, however. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Jeff LaCoursiere wrote: In general you don't need to worry about that, as when you go to buy your routes, the splits are given to you. For example, though you have split up New Zealand nicely I don't need that information, as the termination provider I buy New Zealand from gives me one price for what they deem proper (01164) and another several for what they deem mobile (01164900, 011648, 011642). Whatever destination is dialed simply picks the route that it most matches, and I know what the charges are. Only when it's simple. When a country is small or is big but has a single state telco incumbent and a few mobile carriers, that's not too hard. Of course you can get blended domestic US48 termination - most people do. But, two things happen when you hit a large traffic volume that cause that to go away: 1) *You* lose on the blended rate for high-density and/or highly competitive destinations you could route to for rates under the blended plan. 2) Your providers get increasingly nervous about their exposure to your all-over-the-map traffic patterns, ability to adhere to a theoretical 80% RBOC blend, or whatever. So, high amounts of traffic start to get broken apart into much, much more granular (and therefore numerous) tiers, sometimes down to the terminating carrier. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
In general you don't need to worry about that, as when you go to buy your routes, the splits are given to you. For example, though you have split up New Zealand nicely I don't need that information, as the termination provider I buy New Zealand from gives me one price for what they deem proper (01164) and another several for what they deem mobile (01164900, 011648, 011642). Whatever destination is dialed simply picks the route that it most matches, and I know what the charges are. Case in point (and why we should have a community orientated approach to this) If that is how your carrier has divided it up they have given you inaccurate information. Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Only when it's simple. When a country is small or is big but has a single state telco incumbent and a few mobile carriers, that's not too hard. Of course you can get blended domestic US48 termination - most people do. But, two things happen when you hit a large traffic volume that cause that to go away: 1) *You* lose on the blended rate for high-density and/or highly competitive destinations you could route to for rates under the blended plan. 2) Your providers get increasingly nervous about their exposure to your all-over-the-map traffic patterns, ability to adhere to a theoretical 80% RBOC blend, or whatever. So, high amounts of traffic start to get broken apart into much, much more granular (and therefore numerous) tiers, sometimes down to the terminating carrier. Well, hopefully some people outside of the USA/Canada will assist me with other destinations on the list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. What is your traffic volume such that you are claiming to speak for most people? For genuinely large traffic volumes, it most emphatically _does_ matter. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008, Michael wrote: In general you don't need to worry about that, as when you go to buy your routes, the splits are given to you. For example, though you have split up New Zealand nicely I don't need that information, as the termination provider I buy New Zealand from gives me one price for what they deem proper (01164) and another several for what they deem mobile (01164900, 011648, 011642). Whatever destination is dialed simply picks the route that it most matches, and I know what the charges are. Case in point (and why we should have a community orientated approach to this) If that is how your carrier has divided it up they have given you inaccurate information. Hmm, I looked over your summary again against the route prefixes I just gave and they seem to match. They aren't as detailed, but that isn't important, as long as I can tell a cellular from a landline, which those prefixes do accomplish. I don't really care how accurate they are either, as long as my carrier will honor the prices for the prefixes they have provided me. Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. You have no idea what an uphill battle you will be fighting, and one that is constantly changing. If the idea is to compile all this info to make a master routing list for making purchases, you really don't need to bother. They will be given to you buy your carriers. NANPA is complex, but for purchasing at the wholesale level blended routes are pretty common, which actually makes it one of the simpler ones. Try the Dominican Republic - I currently have over 1200 routes to this small country, and they cannot be any further collapsed... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Hmm, I looked over your summary again against the route prefixes I just gave and they seem to match. They aren't as detailed, but that isn't important, as long as I can tell a cellular from a landline, which those prefixes do accomplish. I don't really care how accurate they are either, as long as my carrier will honor the prices for the prefixes they have provided me. Great! I will send you some 900 calls lol :-) Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. You have no idea what an uphill battle you will be fighting, and one that is constantly changing. If the idea is to compile all this info to make a master routing list for making purchases, you really don't need to bother. They will be given to you buy your carriers. NANPA is complex, but for purchasing at the wholesale level blended routes are pretty common, which actually makes it one of the simpler ones. Try the Dominican Republic - I currently have over 1200 routes to this small country, and they cannot be any further collapsed... Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008 16:45:11 Alex Balashov wrote: Michael wrote: Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. What is your traffic volume such that you are claiming to speak for most people? For genuinely large traffic volumes, it most emphatically _does_ matter. Easy - you are based in the USA, so very likely most of your traffic volume will be in this general area. Where I am based, while there is a lot of traffic volume to North America, there are also large volumes to Pacific and Asia. So therefore the over all USA and NA % is smaller from this part of the world, hence the up line can make enough profit over all that they are less likely to view it as a loosing proposition. There is of course also the psychological element that companies are more polite to people overseas... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008, Michael wrote: Hmm, I looked over your summary again against the route prefixes I just gave and they seem to match. They aren't as detailed, but that isn't important, as long as I can tell a cellular from a landline, which those prefixes do accomplish. I don't really care how accurate they are either, as long as my carrier will honor the prices for the prefixes they have provided me. Great! I will send you some 900 calls lol :-) Which my upstreams will either honor as part of the prefixes they have provided, or will refrain from routing them. I'm not claiming to be anywhere near the top of the foodchain here, and I suppose that yes, I am putting some trust in the carriers I buy from. If I were more paranoid I might try to filter out the possible toll calls and such, but in four years that hasn't been an issue... currently have over 1200 routes to this small country, and they cannot be any further collapsed... Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. Same argument above... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008, Michael wrote: On Sat, 13 Dec 2008 16:45:11 Alex Balashov wrote: Michael wrote: Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. What is your traffic volume such that you are claiming to speak for most people? For genuinely large traffic volumes, it most emphatically _does_ matter. Easy - you are based in the USA, so very likely most of your traffic volume will be in this general area. What he means is that if your traffic volumes get very high you won't be able to purchase blended rates anymore, and then things will get very complicated in NANPA land. If you are REALLY interested in getting started, just pull the rate list from any of a dozen voip terminators and perform the old sort -nu. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: Well, hopefully some people outside of the USA/Canada will assist me with other destinations on the list. That, on the other hand, may not be a bad idea. Although I expect it would become useless in increasing degrees proportional to the level of deregulation and competition in those markets over the next few years. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
You forgot Uruguay I can give you the info if you want :) Enviado desde mi iPhone El 13/12/2008, a las 01:10 a.m., Michael mich...@networkstuff.co.nz escribió: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21 and 29 followed by 6, 7 or 8 digits - Vodafone GSM area code 27 followed by 6 or 7 digits - NZ Telecom CDMA note that there is number portability so the above is a guide. NZ Landline: area code 3, 4, 6, 7 and 9 followed by 7 digits (first digit will be in the range of 2-9) NZ toll free: area code 508 and 800 followed by 6 digits NZ premium: area code 900 - though I doubt any of you will be routing these calls AU cellular: area code 4 followed by a 2 digit network code, and then a 6 digit number Networks include: Optus, Telstra, 3, Vodafone, Virgin and others. All use GSM and there is number portability. AU landline: area code 2, 3, 7 and 8 followed by 8 digits (first digit will be in the range of 2-9) AU toll free: area code 1300 or 1800 followed by 6 digits OR area code 13 followed by 4 digits. AU premium: I'm not sure though someone present may fill us in. Following is the list of countries I need information on: ; ANDORRA ; ARGENTINA ; AUSTRIA ; BAHAMAS ; BELGIUM ; BRAZIL ; BULGARIA ; CANADA ; CHILE ; CHINA ; COLOMBIA ; CROATIA ; CYPRUS SOUTH ; CZECH REPUBLIC ; DENMARK ; ESTONIA ; FRANCE ; GERMANY ; GREECE ; GUADELOUPE ; GUAM ; HONG KONG ; HUNGARY ; ICELAND ; INDONESIA ; IRELAND ; ISRAEL ; ITALY ; JAPAN ; JORDAN ; SOUTH KOREA ; LUXEMBOURG ; MALAYSIA ; MARIANA ISLANDS ; MEXICO ; MONACO ; NETHERLANDS ; NORWAY ; PANAMA ; PERU ; PERU LIMA ; POLAND ; PORTUGAL ; PUERTO RICO ; ROMANIA ; RUSSIA ; SAN MARINO ; SINGAPORE ; SLOVAKIA ; SLOVENIA ; SPAIN ; SWEDEN ; SWITZERLAND ; TAIWAN ; THAILAND ; TURKEY ; UNITED KINGDOM ; UNITED STATES ; VENEZUELA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling e...@fnords.org wrote: Philipp Kempgen wrote: michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? Much like violence and herding of llamas, the r option to Dial (and the Ringing app) almost never solve the problem they are intended to solve and frequently cause more, usually unforeseen, problems. Just say No! to r. If these are inbound calls you are answering, r can be acceptable. But ONLY on the final leg where the call might actually be ringing. But for outbound I fully agree. Let the carrier generate the ringing. A second or 2 of dead air is acceptable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Fri, December 12, 2008 10:50 pm, Michael wrote: Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. It's not that they're letting their underlying carrier do the invoices. It's just that the relationship between them and the underlying carrier is a separate relationship from theirs to you. If they lose money on the rates they guaranteed you, it's their problem, not yours. What's important is what you get charged, not what someone else's cost structure is. It's just that with high traffic volumes it's generally difficult to benefit from that without having the differences proportionally reflected and passed through to you. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Fri, December 12, 2008 10:57 pm, Michael wrote: So therefore the over all USA and NA % is smaller from this part of the world, hence the up line can make enough profit over all that they are less likely to view it as a loosing proposition. That depends entirely on who your users are calling in North America. I know a customer that got a nice blended deal from a Tier 1 NA carrier for terminating traffic from overseas. Said carrier is pulling their hair out trying to figure out how to get rid of this contract; the customer is cherry-picking the most expensive routes off that plan precisely because it is blended. It's costing them hundreds of thousands of dollars a month. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. Some of our providers have rates that don't change much (they've built in tolerance levels to them, so that if there's a fluctuation of 5c in one direction or another, it won't much matter. Some of our providers pass us a new A-Z rate deck every WEEK. Including rate changes and prefix changes. Countries go from 5 prefixes to cover mobile, to 25, and then to 18, and then to 7, and then to 130... changing on a weekly basis (and sometimes daily in a few countries we deal with). You'd need to get more than just the Asterisk community into this. You'd need an overall organisation of underlying carriers worldwide which could update their destinations whenever there's a change. As a project, that's not only daunting technologically, but massively difficult politically. A lot of those UCs aren't going to WANT to join your coalition of information. After all, what's in it for them? Add to that that the information it gives YOU is not going to be applicable on a grand scale. While the actual carrier who maintains prefixes 56-110 may change their structure on a weekly basis, it's possible the contracts they have with providers you'd be using have differing information available to the provider. Which means that just because something in the landscape changes, the rates may not change to you (or might change to YOU, but not to someone who uses a different provider that uses the same UC). I'm not sure I can see the value of a community-driven effort to keep track of things which, by nature, are not applicable to everyone in the community, as we all have our own contracts with our own providers and our own set of rates based on our own conditions of traffic. Perhaps you can explain better the value of the proposition in more detail. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
SIP wrote: As a project, that's not only daunting technologically, but massively difficult politically. A lot of those UCs aren't going to WANT to join your coalition of information. After all, what's in it for them? Not to mention that there are plenty of commercial consultancies, tariff watchers and data aggregators that make very good money selling tiny (but ponderous) subsets of this information in machine-processable format. You'd be cutting into the revenue stream of folks like Telcordia, CCMI, etc. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Alex Balashov wrote: You'd be cutting into the revenue stream of folks like Telcordia, CCMI, etc. ... which, of course, there's nothing wrong with. Just be prepared to witness the awesome power of their fully operational legal battlestation. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: On Sat, 13 Dec 2008 19:29:23 you wrote: Alex Balashov wrote: You'd be cutting into the revenue stream of folks like Telcordia, CCMI, etc. ... which, of course, there's nothing wrong with. Just be prepared to witness the awesome power of their fully operational legal battlestation. This is OTTP (Over the top paranoia) I am talking about information gained from people resident in the various countries. I understand. The problem is that such information is useless and an almost entirely pointless waste of time because in all but the simplest of scenarios (like the kind you mention in Australia and NZ) ordinary people do not have this information, or the information is applicable to them only and not of much intersubjective value. It is exceptionally rare to find a country with just a handful of prefixes, especially outside of the Third World. So, the information would have to come from official sources to be comprehensive. That's problem one. Problem two was summarised by the poster SIP: I'm not sure I can see the value of a community-driven effort to keep track of things which, by nature, are not applicable to everyone in the community, as we all have our own contracts with our own providers and our own set of rates based on our own conditions of traffic. In other words, whereas the distinction between a fixed and mobile call may be of importance in *your* particular pricing arrangements with your suppliers, other people have all sorts of different arrangements. There are blended rates for everything, blended rates for some things, decked rates for some things and blended for others, decked rates by LATA, decked rates by carrier, etc, etc, etc. I'm sure there are rate decks by hemisphere, rate decks by how many goats you are prepared to ritually sacrifice as tribute to ITSP, etc. So, the distinction useful to you in your rating process is not necessarily useful to others, or even a critical mass of others. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Here is the data for Israel: +972 is the international prefix, and then: 2,3,4,8,9 and then 7 digits: Landline, according to the dialling area; usually copper connected phones. 7x and then 7 digits: Landline, country wide numbering (usually IP based operators). x: 2=Golden lines (012), 3=Barak (013), 4=Globecall Partner, 6=Bezeq, 7=HOT. 5x and then 7 digits: Cellulars. x: 0=Pelephone, 2=Cellcom, 4=Orange, 7=MIRS, 9=Jawwal (Palestenian operator). 159 and then 6 digits - Country wide numbering. Used usually by small service providers who has more than one branch (like computer labs, etc.). 17xx and then 6 digits: Usually call centers; local call tariff. 18xx and then 6 digits - Toll free numbers. 19xx and then 6 digits - Premuim services. 10x - Emergency services (police, first aid, fire). 12xx - Special long term services with national importance to have short number (like psychological first aid). This is also used for temporary numbers assigned for short period (like hospital number just after some disaster). * (can also be reached with 1222) - commercial services who want short dialling string. Maybe there are more, but these are the mostly used ones. regards, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
What's more, the official numbering plan authorities' information doesn't provides even less insight into the cost structure once you take into account that different underlying carriers have different ways of delivering calls to different mobile providers. For instance, yes, it is a general rule that termination to mobile providers is more expensive than to fixed-line pretty much anywhere. But what if the underlying carrier isn't delivering it through an incumbent tandem but has a private interconnection (on different terms than standard handoff) with that mobile carrier because they pass enough traffic symmetrically? What if they do bill-and-keep? Now what's it matter? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: There IS life outside of the USA (shock, horror!) I am not making the metaphysical error of assuming otherwise. I'm just pointing out that routing complexity introduces exponentially with competition. Anyone can download NZ's entire numbering plan down to suburb/town level from the NZ Telecom wholesale website. I doubt they would be alone in this. Sure, I just gave you the link to NANPA's equivalent of that. What good is it going to do you, whether with regard to the US or in any other country that has more carriers than thou hast fingers? Because LOTS of countries differenciate between landline and cellular calls. From my list here I would say over 80% do. What do you mean by countries differentiate? Countries don't differentiate anything - carriers differentiate. And the nature of those differences is inextricably bound up in their interconnection agreements. Many international backbone carriers privately interconnect with mobile carriers and bypass the incumbent telco (if the country's laws allow this), allowing them to achieve lower termination rates. The point is that this landline vs. mobile distinction is not particularly universal, not particularly uniform, and therefore, not particularly useful. It's rather specious. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
