RE: [Asterisk-Users] Digium hardware
Hello, The 5 exchange lines I assume they are analogic. For them you will need 5 FXO ports. You can buy a TDM04B and a TDM01B (this will get you to the 5 FXO). Make sure you have 2 PCI slots available. Now for the extensions you need - IP Phones or - ATAs (if you want to reuse your analog phones) IP Phones you can buy from different companies. I like Aastra, Polycom... For ATAs I like Linksys PAP2. Bogdan Moldovan VoIP SIP: sip://[EMAIL PROTECTED] VoIP IAX: iax://obelisk.modulo.ro/101 MODULO Consulting The Future Is Not What It Used To Be http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cisco - Kameko Sent: Wednesday, January 25, 2006 9:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Digium hardware Hello, I want to setup an asterisk pabx. I want to understand more on what hardware (PCI cards) i will need to do this. I have 5 xchange lines and 30 extensions within our offices. I have just finished installing Fedora Core and downloaded asterisk-1.2.3.tar.gz http://ftp.digium.com/pub/asterisk/asterisk-1.2.3.tar.gz and zaptel-1.2.2.tar.gz http://ftp.digium.com/pub/zaptel/zaptel-1.2.2.tar.gz which i want to install. In need or your advise ASAP Regards, SOUL ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to set features.conf to change thehangup key.
If you are on 1.2.1, do:
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is the line you should add or edit
Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Sunday, January 08, 2006 10:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
I have upgraded to Asterisk 1.2.1 and haven't gotten it to work yet.
Does it depend on some options in the Dial command?
I have also got the source now, and would like to know how it can be
modified.
There is no documentation on what the structures do, how which parameters
contain the right settings.
This appears to be the relevant line. What changes does it require to change
the setting?
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
/Obelix
Hello,
The idea is the following:
For the 1.2.1 installation just set the parameter disconnect = *97 In
your features.conf
For the 1.0.7 installation you either upgrade or patch the code. The
patch the code would require you a lot of knowledge of c
programming. It would consist of extracting from the 1.2.1 code the
disconnect functionality and add it to the 1.0.7 code base. But that is
not straight forward...
If you need it badly we can do it for you as consulting. But I
strongly advise you to upgrade.
Upgrade,now, is not an easy task either, but it might be easier that
the code patch. Mainly because you would have to migrate the
configuration or test it... Do you have a test bed?
BR
Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Tuesday, January 03, 2006 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change
thehangup key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
I don't have this in my main installation, which is 1.0.7.
In the case of 1.0.7 where else can I effect that change?
I also have a 1.2.1 setup, what would I have to change in the code below?
What is the general idea?
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] = {
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #,
#, builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, ,
, builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, ,
, builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
In case you do not have this, good changes are that, in case you
need badly this feature, you will upgrade or tweak the sources...
Bogdan
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RE: [Asterisk-Users] Dial(Console/dsp) and option g doesnt appear towork
Have you tried? exten = 309,1,System(echo /tmp/file)exten = 309,2,Dial(Console/dsp,,g)exten = 309,103,System(rm -f /tmp/file) exten = 309,104,Hangup()exten = 309,3,System(rm -f /tmp/file)exten = 309,4,Hangup() Bogdan MoldovanMODULO Consulting"The Future Is Not What It Used To Be"http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry GeisSent: Wednesday, January 04, 2006 8:25 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dial(Console/dsp) and option g doesnt appear towork I have a case where I need the option g to continue execute after the hangup (I'm using 1.2.1)and I have the following in my extensions:exten = 309,1,System(echo /tmp/file)exten = 309,2,Dial(Console/dsp,,g)exten = 309,3,System(rm -f /tmp/file)exten = 309,4,HangupHowever, after the hangup priority 3 is not executed.Does 'g' not work with console/dsp or do I have something wrong.THanks,Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Hello,
The idea is the following:
For the 1.2.1 installation just set the parameter
disconnect = *97
In your features.conf
For the 1.0.7 installation you either upgrade or patch the code. The patch
the code would require you a lot of knowledge of c programming. It would
consist of extracting from the 1.2.1 code the disconnect functionality and
add it to the 1.0.7 code base. But that is not straight forward...
If you need it badly we can do it for you as consulting. But I strongly
advise you to upgrade.
Upgrade,now, is not an easy task either, but it might be easier that the
code patch. Mainly because you would have to migrate the configuration or
test it... Do you have a test bed?
BR
Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Tuesday, January 03, 2006 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
I don't have this in my main installation, which is 1.0.7.
In the case of 1.0.7 where else can I effect that change?
I also have a 1.2.1 setup, what would I have to change in the code below?
What is the general idea?
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] = {
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #,
#, builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
In case you do not have this, good changes are that, in case you need
badly this feature, you will upgrade or tweak the sources...
Bogdan
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RE: [Asterisk-Users] IAXTEL??
From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXTEL??
I know, this is the sad part :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Tuesday, January 03, 2006 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXTEL?? That message has been there for months. On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote: From: http://www.iaxtel.com/ The IAXTel Server is currently under maintenance. Some technical difficulties, such as connection timeouts, registration timeouts, and the inability to make phone calls may be experienced. Thank you for your patience. :( b -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, January 03, 2006 5:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAXTEL?? Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FC3 or FC1 (or something else?)
IMHO use FC4. Also after the install of the OS and all the required packages do a 'yum update'. Bogdan Moldovan MODULO Consulting The Future Is Not What It Used To Be http://www.modulo.ro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Tuesday, January 03, 2006 6:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FC3 or FC1 (or something else?) Hi I wish to install asterisk 1.2 (the latest tar.gz from the site not the CVS version) on an HP box with a TE110P (single port E1/T1) My question is which OS would be preferred in this configuration Fedora Core 1 or Fedora Core 3, and are there any install guides out there that are recent enough for asterisk 1.2 I am also open to suggestions for other Operating Systems if any of you feel that FC1/3 are not the best for the job, my only definates are that I use the latest tar.gz from the asterisk.org website not the CVS and also that I will be using the TE110p Any help would be greatly appreciated Gary ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] offline g.729 transcoding
Hello Kevin, Matt Ridell also replied to a another message with this link (10x both)... But is there a way to do that using a command line like sox? Can sox enc/decode from/to g.729? WIth an external/builtin library? Or something similar to sox? Thanks, Bogdan MoldovanMODULO Consulting"The Future Is Not What It Used To Be"http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin lingSent: Tuesday, January 03, 2006 5:31 AMTo: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding try this: http://www.asteriskguru.com/audio_conversion.php From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim HarrisonSent: Tuesday, January 03, 2006 10:52 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] offline g.729 transcoding I'm trying to get some of the sample asterisk gsm files into a g.729 encoding. Is there an offline way of doing this (without a specialized card?) Can someone point me in the right direction?Thanks,-Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] offline g.729 transcoding
Thanks Kevin, Bogdan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin lingSent: Tuesday, January 03, 2006 8:28 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding FYI: http://redice.krisk.org/ g729: http://www.readytechnology.co.uk/open/ipp-codecs/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bogdan MoldovanSent: Tuesday, January 03, 2006 1:46 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding Hello Kevin, Matt Ridell also replied to a another message with this link (10x both)... But is there a way to do that using a command line like sox? Can sox enc/decode from/to g.729? WIth an external/builtin library? Or something similar to sox? Thanks, Bogdan MoldovanMODULO Consulting"The Future Is Not What It Used To Be"http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin lingSent: Tuesday, January 03, 2006 5:31 AMTo: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] offline g.729 transcoding try this: http://www.asteriskguru.com/audio_conversion.php From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim HarrisonSent: Tuesday, January 03, 2006 10:52 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] offline g.729 transcoding I'm trying to get some of the sample asterisk gsm files into a g.729 encoding. Is there an offline way of doing this (without a specialized card?) Can someone point me in the right direction?Thanks,-Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to set features.conf to change the hangup key.
In features.conf [featuremap] automon = *1 ; One Touch Record atxfer = *2 disconnect = *97 ; this is just an example Bogdan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obelix Sent: Saturday, December 31, 2005 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to set features.conf to change the hangup key. I want to modify features.conf to set a different key to hang up call. Rather than the usual * key. I gather it involves some application map settings etc. Does anyone have a clue? I have read the docs but can hardly find any examples. Regards Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] =
{
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #, #,
builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect, *, *,
builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF },
};
In case you do not have this, good changes are that, in case you need badly
this feature, you will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work with Asterisk 1.07? I tried it and it didn't work
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is just an example
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the hangup
key.
I want to modify features.conf to set a different key to hang up call.
Rather than the usual * key. I gather it involves some application map
settings etc.
Does anyone have a clue? I have read the docs but can hardly find any
examples.
Regards
Obelix
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Of course,
This is what res_features.c does...
B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Is there a means of monitoring the call for that sequence of keys then
hanging up the call if they are detected?
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] = {
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #,
#, builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
In case you do not have this, good changes are that, in case you need
badly this feature, you will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change
thehangup key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work with Asterisk 1.07? I tried it and it didn't
work
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is just an example
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the
hangup
key.
I want to modify features.conf to set a different key to hang up call.
Rather than the usual * key. I gather it involves some application
map settings etc.
Does anyone have a clue? I have read the docs but can hardly find
any examples.
Regards
Obelix
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] Semi-OT: porting numbers away
Depending on the forward type. You could put conditional or un-conditional forwarding. As far as I know some telcos are placing restrictions on conditional forwarding (and that depends on a case by case basis) but for un-conditional forwarding I don't see why there could be a limitation. Bogdan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross C Sent: Friday, December 30, 2005 9:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away Thanks Matt. Are there limitations with call forwarding? For example, with Teliax's pay as you go plan you can have a whole bunch of simultaneous calls (we had 12 going the other day). So say we get 10 or 12 calls on our telco number that forwards to Teliax, is there a limit to the number of forwarded calls going on at once? Or does the telco hand-off the call to Teliax, then the telco is no longer involved in that call? I just don't want call forwarding to defeat the purpose of going with an ITSN or limit my capabilities. Also, do I need to have an actual physical analog line to use call forwarding? I have two numbers that I would like to forward, but I really only need one POTS line that would be used by outgoing stuff (911, credit card machines, etc). So could I have 123-4567 forward to Teliax#987-6543 and 123-4568 forward to Teliax#987-6542, but only have one actual POTS line? Or is this heavily dependent on the telco doing the forwarding? Thanks! -ross -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Friday, December 30, 2005 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away Ross C wrote: Thanks, but I'm looking for information on porting numbers when the current provider holding the numbers goes out of business and is unreachable. Can I get the numbers? The business has had the same phone number for almost 30 years and definitely can't lose the number due to some provider's instability. As most VoIP companies are relatively new and small, I'm a bit skittish about porting these numbers to an ITSN, then that company going out of business and not being able to get my numbers back. How would that work? So use call forwarding from the Telco, forward it to a VoIP DID, if you lose the VoIP DID, change the forwarding to another number. That way you can also keep the PSTN line for emergency calls (despite 911 services being offered by various ITSPs, you are relying on the Internet on site being in top shape). For example, I have seen more companies do something strange (or even participate unknowingly in DDOS attacks) rendering their internet connection useless. While there are workarounds (maintain a good security policy, use QOS, dual networks with router-based traffic control), it never pays to have a customer unhappy (or dead in the case of a missed 911 call). Typically most ITSPs rely on SLAs (Service Level Agreements) from upstream providers which will effectively indemnify them in case of upstream failure, a court case is not really useful in the prevention of the situation. Is one POTS line really so much in the end? We normally route outbound calls first via ourselves, and in the case of network failures, fall back to the customer's PSTN/BRI line. (BRI being quite popular here in Italy). This way they have unlimited outgoing lines and a set number of incoming lines (we typically offer per channel on inbound DIDs). If there is ever any problem with the DID, you can forward the PSTN number back to a cellphone etc. In fact, if I remember correctly NuFone (https://www.nufone.net/) in the USA provides a service whereby they will try to route your number via voip and fallback to an alternate number (ideal if available). Furthermore, NuFone is one of the oldest (if not _the_ oldest) IAX provider and has proven to be one of our most stable providers. If you know what you're doing, NuFone would be my recommendation, if however you need quite a bit of hand holding, I'd either recommend another provider, or exhaustive use of the various Asterisk documentation resources. :) You can never guarantee a company is not going to go under, but when a company provides a good service for an extended period of time, you can feel a little safer. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
RE: [Asterisk-Users] Semi-OT: porting numbers away
This is a possible scenario indeed. But this scenario should be handled by the switches of the telco... bogdan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Friday, December 30, 2005 9:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote: Depending on the forward type. You could put conditional or un-conditional forwarding. As far as I know some telcos are placing restrictions on conditional forwarding (and that depends on a case by case basis) but for un-conditional forwarding I don't see why there could be a limitation. Well they generally like limitations because people sometimes show questionable judgement. A forwards to B, B forwards to A. Call comes in on either and you rapidly exhaust capacity. Sometimes its just lack of knowledge that leads people to do this sometimes they just dont think beforehand. For reasons like these they like to put caps on it, but you can generally get the caps high enough that if you are forwarding from an analog line it shouldnt matter (ie if you need 1000 forwards you need to reevaluate how you are doing this). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conditional CODEC translation
Hello Leandro, Indeed, your problem is a nice one. I do not think this is possible to do this with *. If I am wrong, the list, please correct me... There are two ways of doing that: 1/. would be to have the IP phone have a logic that advertises the preferred codec based on B-number. I do not know of any IP Phones that are able to do that... 2/. would be to have * perform allow/disallow parameters based on the number you have dialed. Both would be interesting... Maybe we will implement this in LoudHush (for the softphone side). Could such a conditional codec be implemented on asterisk in a future version? Bogdan Moldovan MODULO Consulting "The Future Is Not What It Used To Be" http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leandro RzezakSent: Thursday, December 29, 2005 6:03 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Conditional CODEC translation We have a VoIP termination provider that allows g729.We would that internal calls (between our own IP phones) be handled using alaw, and outgoing calls using native forwarded g729 without translation (ie, not using asterisk g729 licenses). We need to avoid translations.WHAT WE HAVE NOW:IP Phone --alaw-- IP PhoneIP Phone --alaw-- Asterisk --g729-- VoIP provider(Phones are configured only to allow alaw and g729, provider is configured only to allow g729; however phones are never using g729)WHAT WE NEED:IP Phone --alaw-- IP PhoneIP Phone --g729-- VoIP providerPlease help me accomplish that.Thank you-- Leandro Rzezak[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
