RE: [asterisk-users] How to Install H323

2006-09-11 Thread Wasif 
Hi,

I want to set chan_h323. If you think this is not the best then please tell
me the setup information of the best one.


Thanks for you reply.




-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Saturday, September 09, 2006 7:40 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE : [asterisk-users] How to Install H323

hello ,

Which channel do you want to set chan_h323 chan_oh323
or chan_ooh323 ?

Harry
--- Wasif [EMAIL PROTECTED] a écrit :

 Hello,
 
 Could anyone tell me how to install/configure H323
 with Asterisk 1.2.11 .
 
 
 Thanks
 
 Wazb
 
 ___
 --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
 







___ 
Découvrez un nouveau moyen de poser toutes vos questions quelque soit le
sujet ! 
Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et
vos expériences. 
http://fr.answers.yahoo.com 

___
--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

[asterisk-users] How to Install H323

2006-09-07 Thread Wasif 
Hello,

Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .


Thanks

Wazb

___
--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

[asterisk-users] Asterisk Billing

2006-08-11 Thread Wasif 
Hello,

Does anyone know  about open source wholesale billing for Asterisk?

Thanks


___
--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

[asterisk-users] Asterisk Voicemail Setup

2006-08-10 Thread Wasif 
Hello,

I am using A2Billing which comes with TrixBox 1.1.1 . I am creating SIP
account from A2Billing for IP Phone. Everything is working fine. What I need
is to assign Voicemail box to every phone, which I think cannot be done
through A2Billing right now. Therefore I need to know some utility or
command or any method through which I can create Voicemail account for IP
phones manually.

Thanks


___
--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

[asterisk-users] codec conversion

2006-08-01 Thread Wasif 
Hello,

What is the best utility to convert GSM files into G729 files for batch
processing.


Thanks

WAzb

___
--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

[asterisk-users] Source Directory of ASterisk

2006-07-28 Thread Wasif 
Hi,


I am using TriBox 1.1.1/Asterisk. I want to know where I can find source
directory of Asterisk in system so I can install Asterisk audio conversion
module (http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw prompts
into g729 prompts. It requires to point Asterisk source Include directory.


Thanks

___
--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

[asterisk-users] No Audio

2006-07-28 Thread Wasif 
Hello,

My DIDs are hitting Directly to Asterisk Machine via SIP G729. From there I
am forwarding call to Cisco 3845 via SIP G729. And from Cisco calls are
terminating to my carriers via H323 G729.

DIDs Asterisk -- Cisco 3845  Carrier
Sip G729 Sip G729  H323 G729

Cisco is capable to convert calls from SIP to H323 and H323 to SIP.
My Problem is when a call hits to my Carrier I get no audio at all. Other
side gets ring but upon answering there is no audio.

Below is the output from CLI. I have installed G729 codec in system and its
working fine; there is no Firewall and NAT implementation in my scenario. 

Called Cisco3800/99
Destroying call '[EMAIL PROTECTED]'
asterisk1*CLI
-- SIP read from X.X.X.X:50974:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: 4169076956 sip:[EMAIL PROTECTED];tag=as2db17260
To: sip:[EMAIL PROTECTED];tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

-- SIP read from X.X.X.X:50974:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: 4169076956 sip:[EMAIL PROTECTED];tag=as2db17260
To: sip:[EMAIL PROTECTED];tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: sip:[EMAIL PROTECTED]:5060
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 261

v=0
o=CiscoSystemsSIP-GW-UserAgent 5381 5741 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 18068 RTP/AVP 18 101
c=IN IP4 X.X.X.X
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port X.X.X.X:18068
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263
|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
-- SIP/Cisco3800-056c is making progress passing it to SIP/5060-08e53008


After few seconds call gets disconnect. 


x.x.x.x  Asterisk Box 
y.y.y.y Cisco 3845 12.3T

[Cisco3845]
;disallow=all
allow=g729
dtmfmode=auto
host=y.y.y.y
insecure=very
sendrpid=yes
type=friend


Thanks



___
--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

[asterisk-users] Getting no Audio with G729

2006-07-27 Thread Wasif 
Hello,

Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/
Asterisk. I have pointed a DID from my carrier via SIP through g729 to
asterisk. Problem is I am not getting any audio even though I am getting
DTMF in asterisk. I am trying to run A2billing with asterisks.

Configuration of carrier is asterisk is:
[abc]
allow=g729
context=c-DID
dtmfmode=auto
host=xxx.xxx.xxx.xxx
insecure=very
sendrpid=yes
type=friend
echo=no

Any suggestions ?

Thanks



___
--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

[asterisk-users] RE: Getting no Audio with G729

2006-07-27 Thread Wasif 
Hi again,

Asterisk was not behind the NAT and I downloaded correct platform of codec.
I solved my problem by changing the prompts into G729 format. And it works
fine now. 

Now I need to know about a utility which can convert all ulaw audio prompts
into g729 prompts in bulk. Or is there any was Asterisk can convert ulaw
prompts to G729 prompts by itself during call.

Thanks ,


-Original Message-
From: Wasif [mailto:[EMAIL PROTECTED] 
Sent: Thursday, July 27, 2006 5:07 PM
To: 'asterisk-users@lists.digium.com'
Subject: Getting no Audio with G729

Hello,

Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/
Asterisk. I have pointed a DID from my carrier via SIP through g729 to
asterisk. Problem is I am not getting any audio even though I am getting
DTMF in asterisk. I am trying to run A2billing with asterisks.

Configuration of carrier is asterisk is:
[abc]
allow=g729
context=c-DID
dtmfmode=auto
host=xxx.xxx.xxx.xxx
insecure=very
sendrpid=yes
type=friend
echo=no

Any suggestions ?

Thanks



___
--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

[asterisk-users] Install H323

2006-07-18 Thread Wasif 
Hello,

I just downloaded Tribox 1.1 having Asterisk 1.2.9.1. I need to have H323
support with asterisk like sip. Please guide me how I can do this.

Thanks

Wazb

___
--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

[Asterisk-Users] Mysql Trixbox

2006-06-28 Thread Wasif 
Hello,


I have installed FreeRadius server on Trixbox Server. My problem is mysql is
not letting FreeRadius to login either locally or remotely. I also insert
proper entries in HOST and USERS tables. But it does not work I always get
ERROR 1045 (28000); Access Denied for user 'root'@'localhost'


Thanks

Wazb

___
--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

[Asterisk-Users] G729 Code

2006-06-28 Thread Wasif 
Hi again,

Could anyone tell me from where I can get non-commercial G729 codec and its
installation procedure for Asterisk?

Thanks



___
--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

[Asterisk-Users] Asterisk Cisco 3800

2006-06-15 Thread Wasif 
Hi,

I have Cisco 3800 12.3(11)T1 . I ask my provider to point all DID sip based
to hit CISCO and from there I take them to Asterisk, both boxes are on
public IP address. I just want to disclose my Cisco IP to vendors.

But somehow DID are not hitting to Asterisk properly. I think there
something worn in Cisco configuration. I am using

dial-peer voice 1 voip
description incoming DID
incoming called-number xx

 dial-peer voice 2 voip
 description Outgoing to Tangerine
 huntstop
 destination-pattern XX
 session target ipv4:xx.xx.xx.xx
 dtmf-relay h245-signal h245-alphanumeric


any idea ?.

Thank



___
--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

[Asterisk-Users] Asterisk as Wholesale

2006-06-12 Thread Wasif 
Hi,

I need to use Asterisk as a switch which can handle wholesale traffic with
billing. Please advice me how I can I implement this.


Thanks

___
--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

[Asterisk-Users] PHP UnixODBC MS SQl 2000

2006-06-07 Thread Wasif 
Hi,

I have Asterisk 12.7.1 installed through [EMAIL PROTECTED] CD. and explicitly I
have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database
which in on Windows 2003 Server on remote location.

I tested connectivity through isql and tsql, both utilities are working
fine. 

I need to access MS SQL 2000 Database through PHP. When I tired to check the
connectivity through a Test PHP file I got following results:

Fatal error: Call to undefined function: odbc_connect() in
/var/www/html/odbctest.php on line 3


By Default PHP was configured with following switches: 
'./configure' '--build=i686-redhat-linux-gnu' '--host=i686-redhat-linux-gnu'
'--target=i386-redhat-linux-gnu' '--program-prefix=' '--prefix=/usr'
'--exec-prefix=/usr' '--bindir=/usr/bin' '--sbindir=/usr/sbin'
'--sysconfdir=/etc' '--datadir=/usr/share' '--includedir=/usr/include'
'--libdir=/usr/lib' '--libexecdir=/usr/libexec' '--localstatedir=/var'
'--sharedstatedir=/usr/com' '--mandir=/usr/share/man'
'--infodir=/usr/share/info' '--cache-file=../config.cache'
'--with-config-file-path=/etc' '--with-config-file-scan-dir=/etc/php.d'
'--enable-force-cgi-redirect' '--disable-debug' '--enable-pic'
'--disable-rpath' '--enable-inline-optimization' '--with-bz2'
'--with-db4=/usr' '--with-curl' '--with-exec-dir=/usr/bin'
'--with-freetype-dir=/usr' '--with-png-dir=/usr' '--with-gd=shared'
'--enable-gd-native-ttf' '--without-gdbm' '--with-gettext'
'--with-ncurses=shared' '--with-gmp' '--with-iconv' '--with-jpeg-dir=/usr'
'--with-openssl' '--with-png' '--with-pspell' '--with-xml'
'--with-expat-dir=/usr' '--with-dom=shared,/usr' '--with-dom-xslt=/usr'
'--with-dom-exslt=/usr' '--with-xmlrpc=shared' '--with-pcre-regex=/usr'
'--with-zlib' '--with-layout=GNU' '--enable-bcmath' '--enable-exif'
'--enable-ftp' '--enable-magic-quotes' '--enable-sockets' '--enable-sysvsem'
'--enable-sysvshm' '--enable-track-vars' '--enable-trans-sid' '--enable-yp'
'--enable-wddx' '--with-pear=/usr/share/pear' '--with-imap=shared'
'--with-imap-ssl' '--with-kerberos' '--with-ldap=shared'
'--with-mysql=shared,/usr' '--with-pgsql=shared' '--with-snmp=shared,/usr'
'--with-snmp=shared' '--enable-ucd-snmp-hack' '--with-unixODBC=shared,/usr'
'--enable-memory-limit' '--enable-shmop' '--enable-calendar' '--enable-dbx'
'--enable-dio' '--enable-mbstring=shared' '--enable-mbstr-enc-trans'
'--enable-mbregex' '--with-mime-magic=/usr/share/file/magic.mime'
'--with-apxs2=/usr/sbin/apxs'


Please guide me what else should I need to do.
 

Thanks

Wazb

___
--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

[Asterisk-Users] Asterisk FAx

2006-04-28 Thread Wasif 
Hi,

I have configured Asterisk with Fax-to-Email feature. Fax is coming to
Asterisk through DID. What is happening is that sometimes Asterisk receives
Fax in first attempt and sometimes in 2 to 4 attempts. On DID Sip,G711 codec
 T.38 protocol is enabled.

Please advise me how I can make Fax service more reliable on Asterisk.

Thanks

Wazb

___
--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

[Asterisk-Users] Callback help

2006-04-26 Thread Wasif 
Hi,


I need to implement international Callback service by using A2billing. Could
anyone guide me where I can find some good material regarding this.

Thanks

Wasif

___
--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

[Asterisk-Users] Asterisk FAx-to-Email

2006-04-21 Thread Wasif 


-Original Message-
From: Wasif [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 20, 2006 4:25 PM
To: 'asterisk-users@lists.digium.com'
Subject: Asterisk FAx-to-Email


Hi,


I get error when my DID hit to asterisk box which I am using for FAX to
Email Service. Sometimes Fax goes through but mostly I get communication
error on Fax Machine and on Asterisk I get Comfort noise support incomplete
in Asterisk (RFC 3389) error.

I am using SIP with G711. My Did provider cannot turn off VAD and Echo from
his side, so is there any option or setting I can do at my side to make FAX
service more reliable



Thanks

Wazb

___
--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

[Asterisk-Users] Asterisk FAX-to-Email

2006-04-21 Thread Wasif 
Hi,

How can we change the FROM address when Asterisk sends mail (in FAX-to-Email
feature). For example it is sending [EMAIL PROTECTED] in FROM
address; I need to change it to [EMAIL PROTECTED] 

Any help?



Wazb

___
--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

[Asterisk-Users] Asterisk FAX

2006-04-20 Thread Wasif 
Hi,

How can we change the FROM address when Asterisk sends mail. For example it
is sending [EMAIL PROTECTED] in FROM , I need to change to
[EMAIL PROTECTED] 

Any help?

Thanks


Wazb

___
--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

[Asterisk-Users] Asterisk FAx-to-Email

2006-04-20 Thread Wasif 

Hi,


I get error when my DID hit to asterisk box which I am using for FAX to
Email Service. Sometimes Fax goes through but mostly I get communication
error on Fax Machine and on Asterisk I get Comfort noise support incomplete
in Asterisk (RFC 3389) error.

I am using SIP with G711. My Did provider cannot turn off VAD and Echo from
his side, so is there any option or setting I can do at my side to make FAX
service more reliable



Thanks

Wazb

___
--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

[Asterisk-Users] Asterisk (RFC 3389)

2006-04-20 Thread Wasif 
Hi,


I am getting this message when my DID hit to asterisk box  which I am using
for FAX to Email Service.

Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible

Any cure for that.


Thanks

Wazb

___
--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

[Asterisk-Users] NAT/STUN Server

2006-04-13 Thread Wasif 
Hi,

I am trying to register SIP clients which are behind NAT on different
network. In order to achieve this goal I think I need STUN Server . I
downloaded STUN Server from
http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz
 But I don't know how to install/configure it. 

And please advice me that STUN server is good idea for this scenario?

Thanks in advance

Wazb

___
--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

[Asterisk-Users] STUN Server info

2006-04-11 Thread Wasif 
Hi,

Do we need STUN server with Asterisk(1.2.6) for SIP phones which are using
NAT on different networks ???


Thanks

___
--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

[Asterisk-Users] Need to Install Fax to Email feature

2006-04-06 Thread Wasif 
Hi,

I need to receive FAX over DID and forward that FAX in email to particular
person. I read some articles about www.voip-info.org but I am confused in
HylaFax, IAXmodem  spandsp. 

Can anyone guide me what is what and how can I achieve my goal.

Thanks

___
--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

[Asterisk-Users] Need to Install Fax to Email feature

2006-04-03 Thread Wasif 
Hi,

Can anyone tell me where I can get Spandsp for Redhat 9. and any good link
to enable Fax to Email feature.

Thanks

___
--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