similar to: Number portability in other parts of the world.

Displaying 20 results from an estimated 10000 matches similar to: "Number portability in other parts of the world."

2009 Jun 09
5
IAX2 issue?
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to the US. The IP address of the remote end changed (though in the config file it's registered as a name i.e. asterisk.remote.end), my system didn't recognised the IP change, it must be cached once and then the cached value used for ever. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857
2005 Jan 09
5
Little confused about Caller ID
OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)
2010 Aug 02
4
Femtocell to VoIP?
Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm
2006 Dec 02
3
Problem in Poland
Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex ____________________________________________________________________________________ Do you Yahoo!? Everyone is
2006 May 01
12
CallerID Name problem
I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !
2011 Nov 03
15
DID from Direct from Telco
Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can
2007 May 12
2
zonedata.c
Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm
2006 Apr 13
3
Will VoIP ITSP's be Next?
Will VoIP be Next? Telco's that provide Internet services to their customers are now trying to charge select companies for large volumes of content that pass over their network to their paying customers! What part of this "greed fest' makes any sense to you? Telco's sell DSL telling customers how much faster it is, how much they can do with Highspeed Internet connections and now
2004 Apr 21
6
Help choosing a UK IAX provider
Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm
2008 May 14
3
Question about SS7
Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of
2007 Apr 24
4
Marketing 101
I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.
2006 Apr 22
4
Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding
2008 Oct 01
3
GSM / 3g channel bank
More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30
2006 Jan 29
4
How to remove first ring tone on FXO?
Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Dec 13
6
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
2006 May 22
10
US telco lingo
Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------
2008 Jul 21
1
"the specified network name no longer exists"
hi all i'm experiencing some really funny behavior on our samba server (CentOS5, Samba Version 3.0.25b-1.el5_1.4) from time to time, our xp/2000 users (our workstations are xp/2000 based) are reporting that while they try to copy a file from one location of the file server to another, they are presented with the the error.... "the specified network name no longer exists" if they
2006 Mar 13
4
priorityjumping=no
I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>
2004 Aug 13
2
Static on outgoing calls using either X100P or TDM400P
Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if
2003 Jul 11
8
G729 codec problems
Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator