similar to: country/city codes

Displaying 20 results from an estimated 1100 matches similar to: "country/city codes"

2005 Feb 11
1
SIP in the Philippines
I have a frend in Manilla who is trying to connect to an Asterisk-based VoIP provider here in Western Canada. Has anyone had difficulties with SIP in the Philippines ? I'm having a lot of trouble getting info from the provider there (PLDT) and it seems as if the device can't access a port that will allow it to get out and REGISTER with the switchboard even because the provider never sees
2003 Sep 23
9
dialing codes..( You can help! )
Hi, I am trying to setup some LCR functions on my Asterisk box and have a cheap call provider that uses various different numbers for landlines and cell phone numbers in various countrys.. I am finding it difficult to find the various codes.. eg. UK Landline - +44[12]. UK Cell - +44[7]. SA Landline - +27[1-6]. SA Cell - +27[78]. Please send me your country's dialing rules similar to how I
2004 Aug 12
9
Asterisk and SER
Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:
2004 Sep 21
2
login from XP
Hello, I'm having a problem logging in to the Samba domain from an XP machine. The machine is added to the domain but after reboot I can't login. I've done the signorseal registry hack but I'm still not able to login. The error I get when logging in is this: "Windows cannot connect to the domain, either because the domain controller is down or otherwise unavailable, or
2005 Feb 03
5
OT: How to "own" a telephone number?
Hello! We are open to the possibility of changing our business telephone number shortly. This will most likely be necessary due to a physical move, changing providers and a few other reasons. However, we woud like this to be the *last* time we need to do this. Ever. No matter what. Is that possible? On the Internet, you get this power with domain names. We "own" our domain
2012 Dec 27
4
How do *you* test your changes to dialplans ruled by GotoIfTime?
This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan
2009 Mar 12
2
Timeout for Queue
Hello, We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code
2017 Apr 18
2
SIP connections over OpenVPN connection get one-way voice.
2000 Sep 25
1
How do I get the username in channel_input_port_open
Hello, Does anyone know how to get the username of the user (already authenticated and logged in) who is sending data to a forwarded tunnel from the channel_input_port_open function in channels.c?? I've tried numerous things, and all I can get is the IP address that is sending the data and where it is going to be sent to. All I want is the username or the UID Here is the chain of events that
2009 Jan 20
1
Setting up an outgoing trunk group
Hi All, I'm confused! My Asterisk system has a Zap trunk and three SIP trunks. I'd like to configure the dialplan to route via the first trunk in a list and if that's not available or it's busy, fall over to the second, then to the third, etc. AIUI Dial(Zap/1&SIP/out1&SIP/out2/${EXTEN}) rings all the trunks in the list and bridges to the first to answer. Unfortunately,
2016 Feb 17
2
Problem compiling res_fax_spandsp.c on Debian server.
On 2016-02-17 15:32, Richard Mudgett wrote: > On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maillist at lightspeed.ca> > wrote: > >> Hi everyone. >> >> We have an Asterisk server running Debian Squeeze, with Asterisk >> v1.8.13.1 (basically, the Debian Stable version for Squeeze, but >> with some minor source code changes specific to our site).
2009 Mar 22
3
I need a country, state, city database
I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=world&v1=l&v2=s&v3=a&pricing=99 Regards, Dean Collins Cognation Inc dean at cognation.net
2010 Aug 20
5
country state city drop down list rails
hi every one please tell me the recommended way to get country state city drop down list in rails any gem - plug-in tutorial thanks in advance. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe
2017 Apr 18
2
SIP connections over OpenVPN connection get one-way voice.
2011 Mar 07
2
Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status
2009 May 26
2
Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn
2010 Nov 12
3
Sending calls to a particular T1 port.
We have two Asterisk servers. One is a live server supporting our customers, and the other is a backup server that's being upgraded and pressed into service. Both servers have a Digium TE405P T1 card in them, and in order to test the T1 service on the backup server, I've created a T1 crossover cable (as per http://www.voip-info.org/wiki/view/crossover+T1+cable) that goes from port 4 on the
2004 Mar 10
1
Map of British Colonial America 1775
Does anyone have suggestions about how to produce a map of British Colonial America 1775? At that time, Great Britain had 26 colonies in the Americas, including Bermuda, several Caribbean islands, "Quebec" (extending then almost to New Orleans), Nova Scotia, Newfoundland, Belize, and the 13 that declared independence in 1776. I've reviewed the "map"
2006 Jun 12
15
Mongrel Now Recommended Setup?
I see that the RubyOnRails.com site has migrated to Mongrel with Apache as a front-end proxy. Is that now the/a recommended setup for Rails apps? We''re still using FCGI but I''m always interested to learn more about other folk''s successful deployment choices. Across all of our sites we''re pumping out about 300k pages per day so anything that saves memory or has
2016 Feb 17
2
Problem compiling res_fax_spandsp.c on Debian server.
Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old