RE: [asterisk-users] How to Install H323
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
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
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
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
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] No Audio
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" ;tag=as2db17260 To: ;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" ;tag=as2db17260 To: ;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: 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] Source Directory of ASterisk
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] RE: Getting no Audio with G729
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] 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
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] G729 Code
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] Mysql Trixbox
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] Asterisk & Cisco 3800
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
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
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
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
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
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-to-Email
-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 (RFC 3389)
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] 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
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] NAT/STUN Server
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
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
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
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