Re: [asterisk-users] Official numbering plan - where to get?
If you have 10 billing plans from different providers, you have for sure at least almost all the data. Use the prefix from the plans to build your own database of prefixes and destinations. Il giorno 23/mar/2012 05:06, Mikhail Lischuk mlisc...@itx.com.ua ha scritto: ** Is it a problem to parse rates from said 10 providers and create database with all their info? Anyways, speaking of this as a service... I have at least 2 clients, who would love such service: some kind of daily (maybe more often) updated database, which automatically normalizes rates and provides output in parseable format. Maybe even that could include some interactive page, for providers which offer cheaper rates for higher call volumes. But of course 100 Euros/month will be too much for such service. AND some kind of integration with Starbilling will make the whole world happy. BR Don Kelly писал 23.03.2012 01:00: Although I do feel that 100+ Euros/month is more than most of us could manage, I don't think a one-time list is of much value. I would be interested in establishing a database if there was interest from enough users for a modest subscription price. --Don Don Kelly PCF Corp People Come First651 842-1000 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Thursday, March 22, 2012 5:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Official numbering plan - where to get? I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx and Http proxies
2012/3/22, John Knight j...@classiccitytelco.com: I've tried this in the past and while FreePBX and its base modules work fine in an http proxy environment, some applications like fop2 fail to connect properly as they obviously rely on direct connections via ajax using the browser as a client. That said, I've never tested the end point manager in this capacity. The error seems to indicate the module doesn't have connectivity when trying to access something on the internet or it's getting a 404 or something. With tcpdump I compared a working (without proxy) and a non-working (with a proxy) Freepbx installations. In the working one, I can see freepbx and mirror.freepbx.org exchanging SYN and ACK packets before the first HTTP GET /provisioning/v2.5/master.xml. In the non-working one, I can see freepbx sending a SYN packet to mirror.freepbx.org and obviously waiting for an answer that do not come. So to me, the End Point Manager module do not care about any of my proxy settings (in /etc/profile, /etc/apache2/envvars and /etc/wgetrc). I took a quick look at source source but could not find anything relevant. Are you able to update and install freepbx modules in a similar manner from the same freepbx installation while it's running behind the proxy? yes That should tell you whether or not your problem is stemming from general connectivity issues and not just the end point module in that regard.Always helps to rule stuff out. I think there is a specific issue within this module. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to add prefix in Extensions.Conf
Hello, I have a DID number 5672531308 , I want to add 92 prefix in it as been told by my provider , so I can I do this in extensions.conf? -- Regards, Muhammad Ali DIDx SUPPORT http://www.didx.net Skype: didxnet Phone: +1-212-655-5763 / +1-850-433-8555 Direct : +1-567-2531308 http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
Hi, How about having 2 NIC cards on the PBX(configure the machine as a gw of sorts). On 3/23/12, Sean McMaster sean.mcmas...@msn.com wrote: Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5
On Mar 22, 2012, at 11:25 PM, Shaun Ruffell wrote: On Fri, Mar 16, 2012 at 11:13:14PM +0400, Vahan Yerkanian wrote: I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with Running dahdi_cfg: DAHDI startup failed: Input/output error Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo? Vahan, Just closing out this public thread. dahdi-linux 2.6.1 will contain what fixed this issue on your machine. I committed onto both trunk [1] and onto the 2.6 branch [2]. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10559 [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10565 Thanks again for your help, SHaun I'd like to publicly thank Shaun for the level of support he provided for the resolution of this issue. Within 10 minutes of my original email he replied in private asking for the ssh access and spent over 4 hours overall just on my machine, over weekend and after-hours, going through the source revisions and eventually finding the problem. I can only wish that all telecom vendors were like this… Thanks Shaun! Best regards, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE
file /usr/lib64/libstdc++.so.6.0.10 /usr/lib64/libstdc++.so.6.0.10: ELF 64-bit LSB shared object, AMD x86-64, version 1 (SYSV), stripped file astdb2sqlite3.o astdb2sqlite3.o: ELF 64-bit LSB relocatable, AMD x86-64, version 1 (SYSV), not stripped file db1-ast/*.a db1-ast/libdb1.a: current ar archive -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Raj Mathur (??? ?) Sent: Thursday, March 22, 2012 9:45 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE On Thursday 22 Mar 2012, Danny Nicholas wrote: Hi gang, I've put 10.X on about 15 different VM's now, but I've run into a buzzsaw on this one and my google-fu has failed me Output of make CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/usr/local/src/asterisk-10.1.3/menuselect' make[1]: `menuselect' is up to date. make[1]: Leaving directory `/usr/local/src/asterisk-10.1.3/menuselect' [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized Try: file astdb2sqlite3.o file db1-ast/*.o file /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so (with the appropriate paths for the first two) and see if they're the same (32 or 64-bit) architecture. The third is likely to be 64-bit anyway, what are the first two? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add prefix in Extensions.Conf
Assuming this is outbound, no problem. For inbound, I don't think so either. Can you be a little more specific? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Ali Sent: Friday, March 23, 2012 4:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to add prefix in Extensions.Conf Hello, I have a DID number 5672531308 , I want to add 92 prefix in it as been told by my provider , so I can I do this in extensions.conf? -- Regards, Muhammad Ali DIDx SUPPORT http://w/ http://w http://ww.didx.net/ ww.didx.net Skype: didxnet Phone: +1-212-655-5763 / +1-850-433-8555 Direct : +1-567-2531308 http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ITSPA 2012 Award for Open Source VoIP Projects
Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projects/ As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy, gateways or media servers, thus the decision of ITSPA for awarding to the group. It was rare when Asterisk was not mentioned as part of the VoIP systems in use by the ITSPA members I spoke to, no surprise! A significant part of the award is therefore (to be paint with) Asterisk logo. As another long time user of Asterisk project, I take the opportunity to send again my thanks to the people behind the project. If anyone is looking for more insights (for news, blogs, personal curiosity) about the event, just drop me an email! Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add prefix in Extensions.Conf
The short answer is yes you can. Now the longer answer is give us more detail if you want to know how. Are they asking you to add the 92 when you dial 5672531308, or is this question really about the callerid number? *** Sam Lutgring Director of Informational Technology Services Calhoun Intermediate school district lutgr...@calhounisd.orgmailto:lutgr...@calhounisd.org www.calhounisd.orghttp://www.calhounisd.org From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Ali Sent: Friday, March 23, 2012 5:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to add prefix in Extensions.Conf Hello, I have a DID number 5672531308 , I want to add 92 prefix in it as been told by my provider , so I can I do this in extensions.conf? -- Regards, Muhammad Ali DIDx SUPPORT http://whttp://w/ww.didx.nethttp://ww.didx.net/ Skype: didxnet Phone: +1-212-655-5763 / +1-850-433-8555 Direct : +1-567-2531308 http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded This email is intended only for the use of the addressee(s) named herein. It may contain legally privileged and confidential information. If you are not the intended recipient, or an authorized representative of the intended recipient, you are hereby notified that any review, copying or distribution of this email and its attachments, if any, is strictly prohibited. If you have received this email in error, please immediately notify the sender by return email and delete this email from your system. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk generating backtrace
Hello, backtrace was created. Can anyone help me with understanding it and telling me what went wrong with my Asterisk-server ? Thanks in advance ! This is Asterisk 1.6.2.22. [root@sip1 ~]# gdb -se /usr/sbin/asterisk -ex bt full -ex thread apply all bt --batch -c core.sip1 /root/backtrace.txt warning: exec file is newer than core file. warning: .dynamic section for /lib64/libssl.so.6 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libcrypto.so.6 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libc.so.6 is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /usr/lib64/libxml2.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libz.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libm.so.6 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libdl.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libpthread.so.0 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib64/libncurses.so.5 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libresolv.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib64/libgssapi_krb5.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib64/libkrb5.so.3 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libcom_err.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib64/libk5crypto.so.3 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/ld-linux-x86-64.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib64/libkrb5support.so.0 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libkeyutils.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libselinux.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libsepol.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/asterisk/modules/res_config_mysql.so is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /usr/lib64/mysql/libmysqlclient.so.15 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libcrypt.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /lib64/libnsl.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/asterisk/modules/app_saycountpl.so is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /usr/lib/asterisk/modules/format_mp3.so is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /usr/lib/asterisk/modules/app_addon_sql_mysql.so is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /usr/lib/asterisk/modules/cdr_addon_mysql.so is not at the expected address (wrong library or version mismatch?) warning: .dynamic section for /lib64/libgcc_s.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff00799000 Cannot access memory at address 0x10007 Cannot access memory at address
[asterisk-users] Silent Monitoring and Meetme
Hi All; If we need the admin to have the ability to hear what the agent is talking without the agent or the customer feel, how this can be done? Is it using MeetMe or something else? How? When the admin start hearing, there will be a peep (because I do not need this, it should be silent monitoring). Also, is there a specific kind IP Phones are required for this? Or any Cisco IP Phones can do this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silent Monitoring and Meetme
Check the chanspy application available in the dial plan. https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_ChanSpy. --- Jayesh On Saturday, March 24, 2012, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; If we need the admin to have the ability to hear what the agent is talking without the agent or the customer feel, how this can be done? Is it using MeetMe or something else? How? When the admin start hearing, there will be a peep (because I do not need this, it should be silent monitoring). Also, is there a specific kind IP Phones are required for this? Or any Cisco IP Phones can do this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adding custom SIP Headers in SIP Response
Hi, I have seen similar post several times on the list but without a proper solution. Is it possible to add custom SIP Headers in 1xx or 2xx response. The SIPAddHeader only works for initial INVITE. Is there any workaround for this. There are certain polycom phones which can open up an URL when a Access-URL SIP Header is sent to it in 200 ok. This can be very useful to make the user's experience interactive with such phones. Any help in achieving this is highly appreciated. Thanks, --- Jayesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Official numbering plan - where to get?
You are welcome to an incomplete dataset I have. Data was gathered from publically available sources, including the ITU and Wikipedia. Data does NOT include information for country code 1. http://rock.nyigc.net/e164.csv.gz Enjoy. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Thursday, March 22, 2012 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Official numbering plan - where to get? I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
And then how will I send calls over to the vpn trunk? -Original Message- From: James Mutuku listmut...@gmail.com Date: Fri, 23 Mar 2012 12:28:26 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN Hi, How about having 2 NIC cards on the PBX(configure the machine as a gw of sorts). On 3/23/12, Sean McMaster sean.mcmas...@msn.com wrote: Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke http://www.agile.co.ke www.zetu.co.ke http://www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
On 24/03/2012 04:49, Sean McMaster wrote: And then how will I send calls over to the vpn trunk? via route with high metric... Regards, Eliezer -Original Message- From: James Mutukulistmut...@gmail.com Date: Fri, 23 Mar 2012 12:28:26 To:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN Hi, How about having 2 NIC cards on the PBX(configure the machine as a gw of sorts). On 3/23/12, Sean McMastersean.mcmas...@msn.com wrote: Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Eliezer Croitoru https://www1.ngtech.co.il IT consulting for Nonprofit organizations eliezer at ngtech.co.il -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
Either give it a 2nd address on the nic that can access the VPN modem You can have lots of addresses on a nic to access different sinners on the LAN Or just make sure the gateway knows to route the ipvpn traffic via the vpn modem Cheers Duncan On 24/03/2012, at 3:55 PM, Eliezer Croitoru elie...@ngtech.co.il wrote: On 24/03/2012 04:49, Sean McMaster wrote: And then how will I send calls over to the vpn trunk? via route with high metric... Regards, Eliezer -Original Message- From: James Mutukulistmut...@gmail.com Date: Fri, 23 Mar 2012 12:28:26 To:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN Hi, How about having 2 NIC cards on the PBX(configure the machine as a gw of sorts). On 3/23/12, Sean McMastersean.mcmas...@msn.com wrote: Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Eliezer Croitoru https://www1.ngtech.co.il IT consulting for Nonprofit organizations eliezer at ngtech.co.il -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
