Re: [asterisk-users] 911 via MAX TNT ??

[Leon Sun]

It should work.


Leon Sun 
Times Telecom 
Tel: 604-279-8787 ext 1586 
Fax: 604-278-2793 
Mobile: 604-780-2668

Click this button now and leave your phone number. Talk to me for free.
powered by www.clicksaya.com
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP..   Would this apply
in that scenario ?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??

Joe,

I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}

Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?

You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.

Hope it can help you


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

Alex..  would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done?  Thank you..
-Joe

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

I believe the ISDN call plan can be configured as part of the trunk
group / route.

Joe Carroll wrote:
 We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???

 PRI DEBUG FOLLOWS:


  --nt SETUP  CRV=14997 (Orig)   Prot=Q931   12:51:47.260 06-06-08
 Bearer_Cap  80 90 A2 (Speech,Rate=64K)
 Channel_Id  A1 83 83 (Pref,Intf=0,Chan=3)
 Calling_Num (National,Restricted,Failed) 229317
 Called_Num  (National) 911

 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
 Sent: Thursday, June 05, 2008 6:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 911 via MAX TNT ??

 Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers.  This PRI is a normal two way circuit that
a carrier would deliver to an end user...



 
 From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
 Sent: Thursday, June 05, 2008 9:27 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] 911 via MAX TNT ??

 On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
 On June 4, 2008 06:20:57 pm Joe Carroll wrote:
 Interestingly enough, on the syslog messages from the TNT we are seeing
 Called = 911, Q850 Cause = 28, SIP Response = 484
 That really looks like the switch that the TNT is talking to is rejecting
the
 number, not the TNT...

 Remember: 9-1-1 is a *dialling pattern*, not a *directory number*;
 it's entirely possible that trunks wouldn't accept it directly.

 This *is* a *LEC* trunk, right?

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink
[EMAIL PROTECTED]
 Designer The Things I Think   RFC
2100
 Ashworth  Associates http://baylink.pitas.com '87
e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727 647
1274

  Those who cast the vote decide nothing.
  Those who count the vote decide everything.
-- (Joseph Stalin)

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--
Alex Balashov
Evariste Systems
Web

Re: [asterisk-users] 911 via MAX TNT ??

[Leon Sun]

Joe,

I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}

Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?

You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.

Hope it can help you


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

Alex..  would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done?  Thank you..
-Joe

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??

I believe the ISDN call plan can be configured as part of the trunk
group / route.

Joe Carroll wrote:
 We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???

 PRI DEBUG FOLLOWS:


  --nt SETUP  CRV=14997 (Orig)   Prot=Q931   12:51:47.260 06-06-08
 Bearer_Cap  80 90 A2 (Speech,Rate=64K)
 Channel_Id  A1 83 83 (Pref,Intf=0,Chan=3)
 Calling_Num (National,Restricted,Failed) 229317
 Called_Num  (National) 911

 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
 Sent: Thursday, June 05, 2008 6:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 911 via MAX TNT ??

 Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers.  This PRI is a normal two way circuit that
a carrier would deliver to an end user...



 
 From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
 Sent: Thursday, June 05, 2008 9:27 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] 911 via MAX TNT ??

 On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
 On June 4, 2008 06:20:57 pm Joe Carroll wrote:
 Interestingly enough, on the syslog messages from the TNT we are seeing
 Called = 911, Q850 Cause = 28, SIP Response = 484
 That really looks like the switch that the TNT is talking to is rejecting
the
 number, not the TNT...

 Remember: 9-1-1 is a *dialling pattern*, not a *directory number*;
 it's entirely possible that trunks wouldn't accept it directly.

 This *is* a *LEC* trunk, right?

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink
[EMAIL PROTECTED]
 Designer The Things I Think   RFC
2100
 Ashworth  Associates http://baylink.pitas.com '87
e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727 647
1274

  Those who cast the vote decide nothing.
  Those who count the vote decide everything.
-- (Joseph Stalin)

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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] How to link 2 existing calls

[Leon Sun]

Hi,

I searched web for few hours and couldn't find any solution about linking 2
calls from Asterisk. This is scenario.

1. A call has been connected from A pstn gateway to my Asterisk waiting with
music.
2. Meanwhile, B call has been connected from B pstn gateway to my asterisk
waiting with music.
3. My asterisk has an application that issues a request to link A call and B
call.
4. Asterisk should issue a re-invite to both A and B gateway and let them
exchange RTP directly. Asterisk should still be working as SIP proxy to
collect signaling(like bye).

Would please anyone suggest how to do step 3 and 4? I wouldn't prefer
conference room type since I like RTP packets go through gateway directly.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadley Rich
Sent: Sunday, August 06, 2006 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ring Groups

On Monday 07 August 2006 06:36, Chris Hembrow wrote:
 I am new to asterisk, and learning as I plod along. Currently, I am
 trying to work out how to create a ring group without using AMP.

You should check out the book - 'Asterisk: The Future of Telephony' - 
published by O'Reilly it's available to buy or download. It will give you a 
good starting point.

 I set my inbound line to ring multiple lines by using
 Dial(SIP/101,SIP/102) but when I answered the call, the lines which
 didn't answer became locked with no dialtone, as though on a call.

That dial line should be Dial(SIP/101SIP/102) - the comma (or a pipe, |) 
separates what to dial from the options to the dial command. typing 'show 
application dial' from the Asterisk CLI will get you all the gory details.

 I thought that setting up a ring group might help, but could only find
 references to creating them through AMP.

'Ring Group' is just an AMP term, you are going about it the right way
above.

HTH

hads

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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RE: [Asterisk-Users] Prices of g729 codec

[Leon Sun]

10$/channel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 02, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Prices of g729 codec

Hi, does anyone know the prices for g729 codecs from Digium? I sent an
email a while ago to them but haven't got any response so far.
Prices are per unit or volume?

Thanks,


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] Prices of g729 codec

[Leon Sun]

You can also build G729 codec by urself via Intel IPP.

Regards



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 02, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Prices of g729 codec

Hi, does anyone know the prices for g729 codecs from Digium? I sent an
email a while ago to them but haven't got any response so far.
Prices are per unit or volume?

Thanks,


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] Looking for advanced consultant services

[Leon Sun]

Hi, there,



At least, Ser + proxy (I
gave Asterisk up) can do large scale for sure by multi proxy server applied. We
can let each proxy server to handle 60-70 calls at same time because its
limited by 100M NIC (150k/s of a 2ways G711U,) amd hundreds calls of g729. Basiclly,
more calls, more servers applied. By using DNS SRV, you can configure as many
servers as you like.



To make our customer happy
with voice and fax, we are using G729 as primary codec for sip call. Low end
computer is enough for media proxy. To handle those G711 calls, P3 server(single
or dual CPU) is enough and is very cheaper from ebay. At our each sites, we are
going to use few Dual PIII 1.1-1.4G server as local media proxy servers.



SER is a proxy and it
should handle huge SIP traffic if you dont take too many database
actions (real time charge and billing, complicated application, etc. for each
call). My idea is just to make ser as easy as a regular SIP proxy.







Regards





Leon Sun









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Sunday, October 16, 2005
10:21 AM
To: Commercial and
Business-Oriented Asterisk Discussion
Cc: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Looking
for advanced consultant services






Hi, 

I have a meeting with an important customer in a
couple of days and I am aware that most of their questions are going to be
related about scability of Asterisk. We want to propose this customer to
integrate Asterisk with SER, but I have a loot of complex doubts that I would
like to known before this meeting. 

I would like to contact with a busines that has
experience with large installations and has already work integrating Asterisk
with Ser. My customer is very worried about NAT Tranversal problematic,
he is thinking on focus the service on SER, so use SIP clients, but he would
like to be able to migrate every user to IAX in a a near future. 

I have questions about a solution that is NAT
Transversal, what beneficits/problems will give me products as JASOMI
(why are better than STUN), STUN installation considerationsetc. Also..
Should I consider SIPFOUNDRY instead SER ? 

If anyone is interested, please send me your hourly rates
as well as details about your implication with large scale proyects, with
Asterisk; SER STUN, etc, so I can evaluate to whom forward my questions (I
do not want to spent time with people who have not enough expertise on this). 
You can contact me at [EMAIL PROTECTED] 

Kind Regards.






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RE: [Asterisk-Users] Canada VOIP provider quality

[Leon Sun]

Try us 



Keywestcommunications.com (wholesale)

Timestelecom.ca (retail)











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical Support
Sent: Tuesday, September 27, 2005
2:43 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Canada
VOIP provider quality







I'm looking at switching VOIP providers, but want to
ensure I move to a company with sufficient capacity.











Can any Canadian VOIP users post/email me with
feedback on their providers?











I'll post the results for all to read..








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RE: [Asterisk-Users] VoIP providers -- California, U.S.

[Leon Sun]

If you want SIP phone PBX hosting or residential partitioning, I can't help.
If you want traffic termination(National and International), we can do it.

Regards

Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jennyw
Sent: Thursday, August 25, 2005 12:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] VoIP providers -- California, U.S. 

Hi,

Just wondering if people could suggest a good VoIP provider that can 
service the San Francisco Bay Area and the Los Angeles area. I've tried 
race.com (recommended to me) but they're kind of hard to get ahold of. 
Any other suggestions? This is for a business, so reliability is key.

I did see the recent thread about this, and while I saw a few mentioned, 
I didn't see anything about how reliable the different vendors are, or 
whether people are using them for business or personal use.

Thanks!

Jen

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RE: [Asterisk-Users] Help with TNT and Asterisk

[Leon Sun]

TNT 11.0.2 with SIP.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin Hunt
Sent: Wednesday, August 10, 2005 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with TNT and Asterisk

I've got it working, but I'm having random echo issues with the TNT.
What TAOS are you running on the TNT ?
Which ethernet card are you using?
When you changed from the default 323 signalling to sip (assuming you
did) did you reboot the TNT ?

W. Kevin Hunt

CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
  

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Brian C. Fertig
 Sent: Wednesday, August 10, 2005 10:03 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Help with TNT and Asterisk
 Importance: High
 
 Im having some problems with connecting a TNT to asterisk.  
 The problem is when the call is sent to asterisk and 
 signaling is done the RTP syncs however no audio is produced. 
  Can someone give me some idea of how to accomplish this?
 
 I am using the standard configs and g711 and 729 do the same. 
  No audio.
 
 
 Public IPs on both ends.  No nat.  Any ideas would be appreciated.
 
 ..o---o.
 Brian Fertig
 NOC/Network Engineer
 Planet Telecom, Inc.
 Tampa, FL Office
 
 
 
 
 This email was scanned by:  Mcafee GroupShield
  CONFIDENTIAL DISCLAMER  All 
 information provided in this email is considered confidential 
 and proprietary of Planet Telecom, Inc. and Telecenter Inc.
 Use of this information by anyone other than the recipient or 
 sender will be considered in breach of agreement.
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RE: [Asterisk-Users] Help with TNT and Asterisk

[Leon Sun]

Set TNT's first codec as g711 in voip {0 0} and media default's voip
profile.

Then try again.

To find real problem, please use ethereal to track all udp packets between
your * and tnt.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C.
Fertig
Sent: Wednesday, August 10, 2005 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with TNT and Asterisk
Importance: High

Im having some problems with connecting a TNT to asterisk.  The problem
is
when the call is sent to asterisk and signaling is done the RTP syncs
however
no audio is produced.  Can someone give me some idea of how to
accomplish this?

I am using the standard configs and g711 and 729 do the same.  No audio.


Public IPs on both ends.  No nat.  Any ideas would be appreciated.

..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office




This email was scanned by:  Mcafee GroupShield
 CONFIDENTIAL DISCLAMER 
All information provided in this email is considered confidential
and proprietary of Planet Telecom, Inc. and Telecenter Inc.
Use of this information by anyone other than the recipient or 
sender will be considered in breach of agreement.
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RE: [Asterisk-Users] inbound caller id name pri - tnt - asterisk

[Leon Sun]

TNT supports caller ID
with any softswitch and any protocol.



Regards















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, August 09, 2005
4:20 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
inbound caller id name pri - tnt - asterisk





The TNT can't pass callerid name as far as I know.









/b











On Aug 9, 2005, at 5:17 PM, Damon Estep wrote:









Anyone
out there have success getting caller id name from a pri, through a lucent tnt,
to asterisk?

What
about from other media gateways?





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RE: [Asterisk-Users] Callback question

[Leon Sun]

http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI+php
http://www.junghanns.net/asterisk/page14.html

make a file as /var/lib/asterisk/agi-bin/callback.agi as following

#!/usr/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(0);
$err=fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$stdlog = fopen('my_agi.log', 'w'); 
while (!feof($in)) {
$temp = str_replace(\n,,fgets($in,4096));
$s = split(:,$temp);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
}
$cf =
fopen(/var/spool/asterisk/outgoing/.$agi[uniqueid].$agi[callerid]..ca
ll,w+); fputs($cf,Channel: Zap/g1/.$agi[callerid].\n);
fputs($cf,Context: callback\n);
fputs($cf,Extension: 604\n);
fputs($cf,SetVar: CALLERIDNUM=.$agi[extension].\n);
fputs($cf,MaxRetries: 3\n);
fputs($cf,RetryTime: 10\n);
fclose($cf);
fclose($in);
fclose($err);
?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Sent: Thursday, August 04, 2005 9:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Callback question

Hi,
I'm interested in a callback feature where I can dial my Asterisk, then 
hangup and Asterisk will call me back and I can then place phone calls or 
whatever I want to do. And also, if I've got voicemail I want Asterisk to 
call me back as well. Are there any scripts for this available?
Any help would be apreciated!
Best regards,
Christian 

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RE: [Asterisk-Users] Callback question

[Leon Sun]

Remember, my sample is for ASTCC. A connection will be connected between
original number and ASTCC. If you need connect 2 numbers, you can find from
those links.

Regards

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Sent: Thursday, August 04, 2005 1:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callback question

Hi Leon,
Many thanks for the links, will try it!
All the best,
Christian


- Original Message - 
From: Leon Sun [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Thursday, August 04, 2005 7:50 PM
Subject: RE: [Asterisk-Users] Callback question


 http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI+php
 http://www.junghanns.net/asterisk/page14.html

 make a file as /var/lib/asterisk/agi-bin/callback.agi as following

 #!/usr/bin/php -q
 ?php
 ob_implicit_flush(true);
 set_time_limit(0);
 $err=fopen(php://stderr,w);
 $in = fopen(php://stdin,r);
 $stdlog = fopen('my_agi.log', 'w');
 while (!feof($in)) {
 $temp = str_replace(\n,,fgets($in,4096));
 $s = split(:,$temp);
 $agi[str_replace(agi_,,$s[0])] = trim($s[1]);
 if (($temp == ) || ($temp == \n)) {
 break;
 }
 }
 $cf =

fopen(/var/spool/asterisk/outgoing/.$agi[uniqueid].$agi[callerid]..ca
 ll,w+); fputs($cf,Channel: Zap/g1/.$agi[callerid].\n);
 fputs($cf,Context: callback\n);
 fputs($cf,Extension: 604\n);
 fputs($cf,SetVar: CALLERIDNUM=.$agi[extension].\n);
 fputs($cf,MaxRetries: 3\n);
 fputs($cf,RetryTime: 10\n);
 fclose($cf);
 fclose($in);
 fclose($err);
 ?


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Christian
 Sent: Thursday, August 04, 2005 9:56 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Callback question

 Hi,
 I'm interested in a callback feature where I can dial my Asterisk, then
 hangup and Asterisk will call me back and I can then place phone calls or
 whatever I want to do. And also, if I've got voicemail I want Asterisk to
 call me back as well. Are there any scripts for this available?
 Any help would be apreciated!
 Best regards,
 Christian

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RE: [Asterisk-Users] TNT and SIP problem

[Leon Sun]

Dave,

Did you enable media profile and put Asterisk IP into proxy field?
Use lines to check you TNT

Read media default
List sip-options



Regards

Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
Sent: Sunday, July 24, 2005 12:54 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TNT and SIP problem


I'm trying to get inbound calls from a TNT working but get 407 errors from 
the TNT. This is what I have in sip.conf:

[maxtnt]
type=friend
host=x.x.x.x
dtmfmode=rfc2833
callerid=MaxTNT maxtnt
context=demo
qualify=yes
disallow=all
allow=g729
allow=ulaw
insecure=very

This is what the TNT is spitting out:

Jul 24 14:55:12 tnt1 1/17: Releasing [EMAIL PROTECTED]: 
Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy 
Authentication Required),Progress Cause = NONE
Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.; 
host 0.0.0.0 [MBID 11; 201-2700674]
Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code 
021

I just have a T1 port from the asterisk machine cabled to the TNT with a 
T1 crossover trying to send calls out of the asterisk machine via T1 and 
back in via SIP until the PRI's are turned up.

dave

-- 
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
   encroachments of those in power than by violent
   and sudden usurpations.- James Madison
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RE: [Asterisk-Users] TNT and SIP problem

[Leon Sun]

Try to use like following

[tnt]
type=friend
context=fromtotnt
dtmfmode=rfc2833 
host=XXX.xxx.xxx.xxx

I am using this way.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
Sent: Monday, July 25, 2005 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TNT and SIP problem


On Mon, 25 Jul 2005, Leon Sun wrote:
 Did you enable media profile and put Asterisk IP into proxy field?
 Use lines to check you TNT
 Read media default
 List sip-options

Yes, my asterisk server is in primary-proxy and registration-proxy. There 
is a trusted-proxy, should that be set? The TNT is still logging 407 
errors, but I have it dumping into a context with a _. extension.

dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
 Sent: Sunday, July 24, 2005 12:54 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] TNT and SIP problem


 I'm trying to get inbound calls from a TNT working but get 407 errors from
 the TNT. This is what I have in sip.conf:

 [maxtnt]
 type=friend
 host=x.x.x.x
 dtmfmode=rfc2833
 callerid=MaxTNT maxtnt
 context=demo
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 insecure=very

 This is what the TNT is spitting out:

 Jul 24 14:55:12 tnt1 1/17: Releasing [EMAIL PROTECTED]:
 Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy
 Authentication Required),Progress Cause = NONE
 Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.;
 host 0.0.0.0 [MBID 11; 201-2700674]
 Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code
 021

 I just have a T1 port from the asterisk machine cabled to the TNT with a
 T1 crossover trying to send calls out of the asterisk machine via T1 and
 back in via SIP until the PRI's are turned up.

 dave



-- 
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
   encroachments of those in power than by violent
   and sudden usurpations.- James Madison
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RE: [Asterisk-Users] 911 Service Providers

[Leon Sun]

We have done it with Group Telecom in Canada but we have to ask customer to
keep their ATAs at fixed place.

Regards

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Oster
Sent: Monday, July 25, 2005 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 911 Service Providers

Who is everyone contracting with for 911 services with the upcoming
FCC deadline?  I've got a few feelers out there working on this issue,
but no real solid leads yet.
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RE: [Asterisk-Users] Asterisk with Lucent TNT echo

[Leon Sun]

No. I never see it.

Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Sent: June 28, 2005 3:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk with Lucent TNT echo

Hey jeremiah,

Do you hear a click click click sound I remember getting that with the
licent tnt with the asterisk server main reason we stopped using the tnt.

Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: carlos at race.com 

-Original Message-
From: Jeremiah Millay [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 28, 2005 2:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk with Lucent TNT echo

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk
server. We have a PRI coming into the Lucent. Basically the problem I'm
having is mostly on inbound calls but some outbound calls as well. I hear
echo and sometimes some weird artifacting on calls coming in from the
lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It
seems like every 3 calls I get is a bad one.
Does anyone on the list run asterisk with this configuration? Does anyone
have any tips to solve this issue?
I've tried modifying the gains at the lucent, as well as turn off and on
jitter buffers on asterisk. Tweaking these seems to help but I'm looking for
something more solid. Any help would be appreciated.
Regards,
Jeremiah

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[Asterisk-Users] SIP NOTIFY message

[Leon Sun]

Hi, All

I would like to send SIP NOTIFY to SIP UA from Asterisk. Is it possible?

I appreciate if you can provide detail sample of message header.


Regards


Leon Sun


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RE: [Asterisk-Users] DID in Western Canada

[Leon Sun]

Nelson,

We do have 780 669  in Edmonton. We also have numbers in Vancouver,
Calgary and Victoria.

CC your reply to [EMAIL PROTECTED] 


Regards


Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nelson Loyola
Sent: June 27, 2005 9:11 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DID in Western Canada

Hello,

I'm having trouble getting finding a company that
provides DID in Western Canada. More specifically in
Edmonton, Alberta. 

We have tried getting in contact with Link2Voip and
Calgary Telecom but neither seems to be answering
their phones or email.

I would appreciate it if anyone can point me in the
right direction.

Thank you,
Nelson

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RE: [Asterisk-Users] Help on installing h323

[Leon Sun]

Go to http://www.inaccessnetworks.com/projects/asterisk-oh323
Download oh323 0.65
Then go to
http://www.inaccessnetworks.com/asterisk-oh323/Libraries
download following

openh323-Janus_patch4-src-tar.gz (2555677 bytes) 
pwlib-Janus_patch4-src-tar.gz (229 bytes)

Please read Readme from 0.65 carefully. You can do it.

Let me know when you still have problem.

Leon Sun


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of craz sead
Sent: June 21, 2005 7:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help on installing h323

Hi all

could somebody help me how to install and setup H323 i
would like to connect asterisk box with huawei/cisco,
but i still dont understand about installing h323 on
asterisk

thaks



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RE: [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show

[Leon Sun]

What kind of toll free do you need? For US only or whole North America?

Do you need carrier send incoming call to your Asterisk by SIP or by T1/E1
from Digium card?


Leon Sun



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken
Sent: June 21, 2005 6:58 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Seeking Inbound 800# Origination for Unique
Prostate Cancer Support Call-In Show

Dear Asterisk Community,
   Does your company provide inbound 800# origination?  If so, please read
this message and e-mail us a quote for monthly co-lo hosting of our
asterisk server and per-minute inbound 800# origination.

The Prostate Cancer Research and Education Foundation (PC-REF) is a
non-profit organization dedicated to helping prostate cancer sufferers and
their loved ones.  We have created a weekly call-in show using Asterisk
that we offer as a FREE service to the public.  Callers can ask their
questions from world reknown experts, or just listen in.  It's kind of
like a talk show except you use your telephone, instead of a radio.

We need a provider who can host our Asterisk Server and provide reliable 
IAX2 or SIP inbound 800# traffic.  The show is one hour per week.  We need
the capability to support 100+ simultaneous callers.  Most callers listen
for the entire duration of the show.

We have been working with another provider for the last several months,
however, after many trials and tribulations, they have determined that
their maximum capacity is 15 simultaneous callers.  They will remain
anonymous for the time being, as I truly believe that they worked very
hard and to the best of their abilities.  However, they were just
technically unable to deliver to our requirements, despite their promises
and best efforts.  As they have been kind enough to offer a complete
refund, I see no reason to embarass them in this forum.  

Therefore, it would be helpful (but not an absolute requirement) for
your company to be able to port/migrate our 800 number so that we can
keep our existing phone number.

We are ready to move quickly and eager to establish a long term, mutually
beneficial working relationship.  This call in show has the potential to
help many Prostate Cancer sufferers!  Your assistance will be recognized
and appreciated!!

Many Thanks,
   -Lee


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RE: [Asterisk-Users] Seeking Inbound 800# Origination for UniqueProstate Cancer Support Call-In Show

[Leon Sun]

Lee,

Please send e-mail to [EMAIL PROTECTED] and give me a call
604 780 2668?


Leon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken
Sent: June 22, 2005 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Seeking Inbound 800# Origination for
UniqueProstate Cancer Support Call-In Show

hi Leon,
   We are initially looking for US only, but eventually would like to add
international toll free numbers.  We would like inbound IAX2 or SIP.

Thanks,
  -Lee



On Wed, 22 Jun 2005, Leon Sun wrote:

 What kind of toll free do you need? For US only or whole North America?
 
 Do you need carrier send incoming call to your Asterisk by SIP or by T1/E1
 from Digium card?
 
 
 Leon Sun
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken
 Sent: June 21, 2005 6:58 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Seeking Inbound 800# Origination for Unique
 Prostate Cancer Support Call-In Show
 
 Dear Asterisk Community,
Does your company provide inbound 800# origination?  If so, please read
 this message and e-mail us a quote for monthly co-lo hosting of our
 asterisk server and per-minute inbound 800# origination.
 
 The Prostate Cancer Research and Education Foundation (PC-REF) is a
 non-profit organization dedicated to helping prostate cancer sufferers and
 their loved ones.  We have created a weekly call-in show using Asterisk
 that we offer as a FREE service to the public.  Callers can ask their
 questions from world reknown experts, or just listen in.  It's kind of
 like a talk show except you use your telephone, instead of a radio.
 
 We need a provider who can host our Asterisk Server and provide reliable 
 IAX2 or SIP inbound 800# traffic.  The show is one hour per week.  We need
 the capability to support 100+ simultaneous callers.  Most callers listen
 for the entire duration of the show.
 
 We have been working with another provider for the last several months,
 however, after many trials and tribulations, they have determined that
 their maximum capacity is 15 simultaneous callers.  They will remain
 anonymous for the time being, as I truly believe that they worked very
 hard and to the best of their abilities.  However, they were just
 technically unable to deliver to our requirements, despite their promises
 and best efforts.  As they have been kind enough to offer a complete
 refund, I see no reason to embarass them in this forum.  
 
 Therefore, it would be helpful (but not an absolute requirement) for
 your company to be able to port/migrate our 800 number so that we can
 keep our existing phone number.
 
 We are ready to move quickly and eager to establish a long term, mutually
 beneficial working relationship.  This call in show has the potential to
 help many Prostate Cancer sufferers!  Your assistance will be recognized
 and appreciated!!
 
 Many Thanks,
-Lee
 
 
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RE: [Asterisk-Users] Bill seconds

[Leon Sun]

The easiest way is to change another vendor asap. It is ridiculous that your
carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter and
billing unit does.


Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: June 15, 2005 10:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds

I've done a little thinking on this one  If you are using ASTCC, it 
would be fairly straightforward to edit it and have it make a 2 second 
adjustment.  If your using another solution it probably would be fairly 
easy also...

Darren Wiebe
[EMAIL PROTECTED]

Americo Sanchez C. wrote:


 Hi all,

 We've installed Asterisk on a rural development project and we're
 testing a prepaid phone service. As far as now we're having terrific
 service results but there's a problem with the calls billing at our
 local telecom. For instance, a farmer buys a 1 dollar phone card and use
 it to dial a USA number, the call should lasts for 60 seconds. Asterisk
 is doing a great job finishing the call exactly at 60 seconds. The
 problem is that the telecom company billing system adds a two second
 delay for each call, so the bill is not for 1 but 2 minutes (they round
 fractions up).

 We're loosing money and the local telecom doesn't seem to have a
 solution for this matter.

 Have you experienced something similar? Do you have any idea of how can
 we solve this? Is it possible to configure Asterisk so that the system
 thinks that a minute has 58 seconds instead of 60?

 _
 MSN Amor: busca tu  naranja http://latam.msn.com/amor/

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RE: [Asterisk-Users] Bill seconds

[Leon Sun]

If you need a SIP 30+6 a-z carrier, let me know. We may do 6+6 for you.

Leon Sun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: June 15, 2005 10:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds

I've done a little thinking on this one  If you are using ASTCC, it 
would be fairly straightforward to edit it and have it make a 2 second 
adjustment.  If your using another solution it probably would be fairly 
easy also...

Darren Wiebe
[EMAIL PROTECTED]

Americo Sanchez C. wrote:


 Hi all,

 We've installed Asterisk on a rural development project and we're
 testing a prepaid phone service. As far as now we're having terrific
 service results but there's a problem with the calls billing at our
 local telecom. For instance, a farmer buys a 1 dollar phone card and use
 it to dial a USA number, the call should lasts for 60 seconds. Asterisk
 is doing a great job finishing the call exactly at 60 seconds. The
 problem is that the telecom company billing system adds a two second
 delay for each call, so the bill is not for 1 but 2 minutes (they round
 fractions up).

 We're loosing money and the local telecom doesn't seem to have a
 solution for this matter.

 Have you experienced something similar? Do you have any idea of how can
 we solve this? Is it possible to configure Asterisk so that the system
 thinks that a minute has 58 seconds instead of 60?

 _
 MSN Amor: busca tu  naranja http://latam.msn.com/amor/

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RE: [Asterisk-Users] Asterisk and Max TNT

[Leon Sun]

Not exactly understand your question. Is it TNT supposed to terminate calls
generated from Asterisk? If so, please attach your VOIP and T1 file or your
extracted media file if your are using SIP.

I am using MAX TNT as SIP gateway with Asterisk and I have given up Digium
card since TNT can do perfect what I want like routing, codec, termination
and origination.

Regards

  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Baird
Sent: June 15, 2005 7:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk and Max TNT

Hello, I'm currently testing Asterisk over a T1 cross connect to a
MaxTNT chassis that we have. It is working fine switching the calls
through, but there is about a 10 second delay from the time Asterisk
initiates the call until the TNT accepts it. It appears to be a ANI
issue, I've changed several settings and formatting options on the T1
between the two, as well as turning on/off the callerid options in
Zapata.conf, it's very strange. I'm pretty sure this is an
interoperability issue between the two devices, I'm looking for a magic
setting. The TNT doesn't have this problem via SIP.

Regards
Michael Baird

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RE: [Asterisk-Users] 1-800 DID in Alberta

[Leon Sun]

Group Telecom and Telus.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: June 15, 2005 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 1-800 DID in Alberta

Are there any 800 DID number providers for Alberta?

-- 
#Joseph
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RE: [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation

[Leon Sun]

Use Adtran Atlas 800. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of qrss
Sent: June 13, 2005 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 multiplexer (or ?) for failover in large
installation

The box that you are talking about sounds a lot like a DACS.  You might
google around on that term to see if any might have automatic failover.  A
DACS can be reconfigured to cross-connect various DS0s on the fly -
although, no matter how fast the switchover, the carrier will always see
that something happened.  Depending upon what type of signalling you are
using, the trunks could end up out of service for a period longer than it
takes to switch to the backup server.  Also, the system can potentially
fail in several ways.  At the T1 level, at the trunk level and at the
application level.  Depending upon the nature of the failure, the one-box
does it all solution seems unlikely to work - at least not by itself.

-Original Message-
From: Mike
Sent: Mon, June 13, 2005 11:35 am

Hi,

Please forgive my terminology, still a bit new to T1s and such.

I'm looking for a way to have 5 T1s from a carrier terminate into some
 type
of box (multiplexer?), then be able to plug 7 asterisk servers into that
 box
(each with single port T1 card) and be able to have 2 * servers go down at
any given time and not actually have the carrier see that anything has
 happened.
Obviously if a * server crashes the calls on it at the time will drop, but
then once the box (multiplexer?) sees that a T1 is down (between the box
and asterisk) it will terminate those DS0's on another T1. Basically some
type of hunting/pooling/load balancing.

Anyone heard of anything like this? Or am I off my rocker?

Thanks,
Mike
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RE: [Asterisk-Users] MCI vs. XO/Allegiance

[Leon Sun]

I prefer MCI since we use their pri and internet. MCI's support is very pro.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Coulson
Sent: June 13, 2005 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance



Wiley Siler wrote:
 Anyone out there using ISDN PRI from either MCI or XO/Allegiance? 

We have a DS-3 full of PRI from X/O. They work great, mostly, but their
tech support sucks. They screw up number ports all the time and about
every week there is some local number I can't dial to via XO which once
I open a ticket mysteriously gets fixed without a good explanation.
Eventually everything works, but you have to beat on them continously to
get things done.

Better than dealing with SBC though.

David

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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Leon Sun]

Not really true about T1 description. When you apply for T1, you need tell
vendor if it's channelized or non-ch. If you are going to use it for 1.5M
network, you need use unchannelized T1. 

  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: June 13, 2005 6:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

Hi David,

  You are correct, I always get those 2 confused. Thanks for the clearing.

Nir S

David Coulson wrote:

Nir Simionovich wrote:
  

 Now, E1 and T1 lines are based upon a channel based connection, which
means you get a line
with X number of data lines and a single control/signalling line. On T1
it means that you have 23
lines dedicated for Voice/Data (each is 64kbps) and a single signaling
line (64kbps). 



A T1 has no seperate signaling line - You're thinking of PRI. T1 gives
you 24 DS0 (64kbit) channels, which you can do whatever you want with.
PRI just shanks off one channel for D channel signaling.

David

  



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