[asterisk-users] Please reply..Not able to call H323 using SIP client
Hi all, I have configured my asterisk server as gateway and gatekeeper both. I am trying to call using SIP agent to h.323 agent but it is not successful. I have configured ooh323.conf as gateway=yes gatekeeper=10.17.112.12 Still its not working. What configuration file I need to change for connecting SIP to H.323? Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not able to call H.323 client by SIP client
Hi all, I have configured my asterisk server as gateway and gatekeeper both. I am trying to call using SIP agent to h.323 agent but it is not successful. I have configured ooh323.conf as gateway=yes gatekeeper=10.17.112.12 Still its not working. What configuration file I need to change for connecting SIP to H.323? Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Communication between two asterisk server
Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Communication between two asterisk server
Hi Bhrugu , Thanks for the reply. I will check it off. Regards, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta Sent: Fri 2/15/2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Communication between two asterisk server hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help in communicating H323 and SIP
Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=1 tcpEnd=2 ;UDP udpStart=1 udpEnd=2 [general] port = 1720 bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. These flags ; are used in the generation of call detail records. ; ;amaflags = default ; ; You may specify a default account for Call Detail Records in addition ; to specifying on a per-user basis ; ;accountcode=lss0101 ; ; You can fine tune codecs here using allow and disallow clauses ; with specific codecs. Use all to represent all formats. ; ;disallow=all ;allow=all ; turns on all installed codecs ;disallow=g723.1; Hm... Proprietary, don't use it... ;allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw ; see doc/rtp-packetization for framing options ; ; User-Input Mode (DTMF) ; ; valid entries are: rfc2833, inband ; default is rfc2833 ;dtmfmode=rfc2833 ; ; Default RTP Payload to send RFC2833 DTMF on. This is used to ; interoperate with broken gateways which cannot successfully ; negotiate a RFC2833 payload type in the TerminalCapabilitySet. ; ; You may also specify on either a per-peer or per-user basis below. ;dtmfcodec=101 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; IP address or Host name - The acutal IP address or hostname of your GK gatekeeper = DISABLE ;gatekeeper=10.142.17.68 ; ; ; Tell As terisk whether or not to accept Gatekeeper ; routed calls or not. Normally this should always ; be set to yes, unless you want to have finer control ; over which users are allowed access to Asterisk. ; Default: YES ; ;AllowGKRouted = yes ; ; When the channel works without gatekeeper, there is possible to ; reject calls from anonymous (not listed in users) callers. ; Default is to allow anonymous calls. ; ;AcceptAnonymous = yes ; ; Optionally you can determine a user by Source IP versus its H.323 alias. ; Default behavour is to determine user by H.323 alias. ; ;UserByAlias=no ; ; Default context gets used in siutations where you are using ; the GK routed model or no type=user was found. This gives you ; the ability to either play an invalid message or to simply not ; use user authentication at all. ; ;context=default ; ; Use this option to help Cisco (or other) gateways to setup backward voice ; path to pass inband tones to calling user (see, for example, ; http://www.cisco.com/warp/public/788/voip/ringback.html https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html ) ; ; Add PROGRESS information element to SETUP message sent on outbound calls ; to notify about required backward voice path. Valid values are: ; 0 - don't add PROGRESS information element (default); ; 1 - call is not end-end ISDN, further call progress information can ;possibly be available in-band; ; 3 - origination address is non-ISDN (Cisco accepts this value only); ; 8 - in-band information or an appropriate pattern is now available; ;progress_setup = 3 ; ; Add PROGRESS information element (IE) to ALERT message sent on incoming ; calls to notify about required backwared voice path. Valid values are: ; 0 - don't add PROGRESS IE ( default); ; 8 - in-band information or an appropriate pattern is now available; ;progress_alert = 8 ; ; Generate PROGRESS message when H.323 audio path has established to create ; backward audio path at other end of a call. ;progress_audio = yes ; ; Specify how to inject non-standard information into H.323 messages. When ; the channel receives messages with tunneled information, it automatically ; enables the same option for all further outgoing messages independedly on ; options has been set by the configuration. This behavior is required, for ; example, for Cisco CallManager when Q.SIG tunneling is enabled for a ; gateway where Asterisk lives. ; The option can be used multiple times, one option per line. ;tunneling=none ; Totally disable tunneling (default) ;tunneling=cisco; ; Enable Cisco-specific tunneling ;tunneling=qsig ; Enable tunneling via Q.SIG messages ; ;-- JITTER BUFFER
[asterisk-users] (no subject)
Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=1 tcpEnd=2 ;UDP udpStart=1 udpEnd=2 [general] port = 1720 bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. These flags ; are used in the generation of call detail records. ; ;amaflags = default ; ; You may specify a default account for Call Detail Records in addition ; to specifying on a per-user basis ; ;accountcode=lss0101 ; ; You can fine tune codecs here using allow and disallow clauses ; with specific codecs. Use all to represent all formats. ; ;disallow=all ;allow=all ; turns on all installed codecs ;disallow=g723.1; Hm... Proprietary, don't use it... ;allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw ; see doc/rtp-packetization for framing options ; ; User-Input Mode (DTMF) ; ; valid entries are: rfc2833, inband ; default is rfc2833 ;dtmfmode=rfc2833 ; ; Default RTP Payload to send RFC2833 DTMF on. This is used to ; interoperate with broken gateways which cannot successfully ; negotiate a RFC2833 payload type in the TerminalCapabilitySet. ; ; You may also specify on either a per-peer or per-user basis below. ;dtmfcodec=101 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; IP address or Host name - The acutal IP address or hostname of your GK gatekeeper = DISABLE ;gatekeeper=10.142.17.68 ; ; ; Tell As terisk whether or not to accept Gatekeeper ; routed calls or not. Normally this should always ; be set to yes, unless you want to have finer control ; over which users are allowed access to Asterisk. ; Default: YES ; ;AllowGKRouted = yes ; ; When the channel works without gatekeeper, there is possible to ; reject calls from anonymous (not listed in users) callers. ; Default is to allow anonymous calls. ; ;AcceptAnonymous = yes ; ; Optionally you can determine a user by Source IP versus its H.323 alias. ; Default behavour is to determine user by H.323 alias. ; ;UserByAlias=no ; ; Default context gets used in siutations where you are using ; the GK routed model or no type=user was found. This gives you ; the ability to either play an invalid message or to simply not ; use user authentication at all. ; ;context=default ; ; Use this option to help Cisco (or other) gateways to setup backward voice ; path to pass inband tones to calling user (see, for example, ; http://www.cisco.com/warp/public/788/voip/ringback.html https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html ) ; ; Add PROGRESS information element to SETUP message sent on outbound calls ; to notify about required backward voice path. Valid values are: ; 0 - don't add PROGRESS information element (default); ; 1 - call is not end-end ISDN, further call progress information can ;possibly be available in-band; ; 3 - origination address is non-ISDN (Cisco accepts this value only); ; 8 - in-band information or an appropriate pattern is now available; ;progress_setup = 3 ; ; Add PROGRESS information element (IE) to ALERT message sent on incoming ; calls to notify about required backwared voice path. Valid values are: ; 0 - don't add PROGRESS IE ( default); ; 8 - in-band information or an appropriate pattern is now available; ;progress_alert = 8 ; ; Generate PROGRESS message when H.323 audio path has established to create ; backward audio path at other end of a call. ;progress_audio = yes ; ; Specify how to inject non-standard information into H.323 messages. When ; the channel receives messages with tunneled information, it automatically ; enables the same option for all further outgoing messages independedly on ; options has been set by the configuration. This behavior is required, for ; example, for Cisco CallManager when Q.SIG tunneling is enabled for a ; gateway where Asterisk lives. ; The option can be used multiple times, one option per line. ;tunneling=none ; Totally disable tunneling (default) ;tunneling=cisco; ; Enable Cisco-specific tunneling ;tunneling=qsig ; Enable tunneling via Q.SIG messages ; ;-- JITTER BUFFER
[asterisk-users] How to register h323 users?
Hi, I have to register h323 users in Asterisk. Please help me in finding out which configuration file to configure. Will it work with gnugk? I am using SJphone as softphone. I am calling users after registering there in the phone and using the ip address of the other system. But I did not register any user in h323.conf or ooh323.conf. Please help me regarding this. Thanking you, With regards, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Could not find ooh323.conf
Hi, I installed Asterisk, asterisk-addons, pwlib, h323plus,opal and gnugk. I am searching for /etc/asterisk/ooh323.conf. It is not there. Can anybody please tell me how to get ooh323.conf. Thanking you, Regards, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Addons install success-Could not find ooh323.conf
Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing something wrong? What I should do to get ooh323.conf Thanking you, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Facing problem in installing asterisk-addons
Hi, I have installed GNU gatekeeper. Then I am trying to install asterisk addons. I gave make and then make clean. I worked properly. Then I gave make install. It gave following error. make[1]: Entering directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 Please help me in understanding the solution for this. Thanking you, Regards, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do Asterisk requires audio codec to be installed?
Hi, Can you please tell me whether Asterisk requires any audio or video codec to be installed separately or it supports itself? Thanking you, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using x-lite -Call failed 404 not found
Actually I registered two users in my X-lite. Both the users registered in different asterisk servers. While calling, first you have to right click on the x-lite and the click on the required server. Then make call. It will work. -Original Message- From: [EMAIL PROTECTED] on behalf of Vincent Sent: Mon 1/28/2008 12:40 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using x-lite -Call failed 404 not found On Mon, 28 Jan 2008 12:01:43 +0530, [EMAIL PROTECTED] wrote: I have installed asterisk.When I start asterisk it starts normally and shows the status running. My partner also installed asterisk. I registered 1 user of her server and 1 user of my server in X-lite. I am able to call or receive call from the users registered in her server but not in my server. Its giving error call failed 404 not found I got the solution. Which was? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installation of gatekeeper-H323plus
Hi, I am trying to install Gatekeeper. I have installed pwlib and trying to install h323 plus. I have set the path as PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/open323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH I also configured /etc/ld.so.conf include /root/pwlib/lib include /root/h323plus/lib Then I tried running ldconfig. But it gave AVC access denied Then I gave # ./configure #make opt Its giving error. /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make[1]: *** [/root/h323plus/lib/libh323_linux_x86_r.so.1.19-beta7] Error 1 make[1]: Leaving directory `/root/h323plus/src' make: *** [opt] Error 2 Please help me out. Thanking you, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using x-lite -Call failed 404 not found
Hi all, I have installed asterisk.When I start asterisk it starts normally and shows the status running. My partner also installed asterisk. I registered 1 user of her server and 1 user of my server in X-lite. I am able to call or receive call from the users registered in her server but not in my server. Its giving error call failed 404 not found What may be the problem? Is my asterisk server not working? How to rectify this problem? Please help me out. Thanking you, Preeta pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using x-lite -Call failed 404 not found
Thanks to all. I got the solution. -Original Message- From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Mon 1/28/2008 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Using x-lite -Call failed 404 not found Hi all, I have installed asterisk.When I start asterisk it starts normally and shows the status running. My partner also installed asterisk. I registered 1 user of her server and 1 user of my server in X-lite. I am able to call or receive call from the users registered in her server but not in my server. Its giving error call failed 404 not found What may be the problem? Is my asterisk server not working? How to rectify this problem? Please help me out. Thanking you, Preeta pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Hi, I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting error AVC access denied. Its saying I need to disable SELinux protection. I do not know what to do. Please help me out. Thanking you, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Gopal krishnan Sent: Fri 1/25/2008 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk Hi Pandey, What type of OS you are using, is it redhat or fedora. and install with latest version. On Jan 25, 2008 9:55 AM, Lyle Giese [EMAIL PROTECTED] wrote: You need to do a 'make' before the 'make install'. Lyle [EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provide a proper link to download Libpri-1.4.3
Hi, I tried to install Libpri-1.4.3 after downloading from sites- www.asterisk.com and www.downloads.digium.com. But in both the case the problem is coming AVC access denied. I am using Fedora core 8. I asked this problem earlier and got advice to disable SELinux. But many people adviced not to do this as it does not require and if it is demanding then there is a bug. I am very much confused. I am very new to Asterisk. Please help me. Thanking you, Preeta pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Finding difficulty in installing Asterisk
Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o pri.o pri.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o q921.o q921.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o prisched.o prisched.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o q931.o q931.c In file included from q931.c:28: pri_internal.h:263: error: expected declaration specifiers or ... before size_t q931.c: In function receive_calling_party_number: q931.c:949: error: too many arguments to function libpri_copy_string q931.c: In function transmit_keypad_facility: q931.c:1425: error: too many arguments to function libpri_copy_string q931.c: In function q931_keypad_facility: q931.c:2492: error: too many arguments to function libpri_copy_string q931.c: In function pri_release_finaltimeout: q931.c:2667: error: too many arguments to function libpri_copy_string q931.c: In function q931_setup: q931.c:2817: error: too many arguments to function libpri_copy_string q931.c:2820: error: too many arguments to function libpri_copy_string q931.c:2837: error: too many arguments to function libpri_copy_string q931.c:2854: error: too many arguments to function libpri_copy_string q931.c:2860: error: too many arguments to function libpri_copy_string q931.c: In function q931_receive: q931.c:3314: error: too many arguments to function libpri_copy_string q931.c:3315: error: too many arguments to function libpri_copy_string q931.c:3316: error: too many arguments to function libpri_copy_string q931.c:3318: error: too many arguments to function libpri_copy_string q931.c:3319: error: too many arguments to function libpri_copy_string q931.c:3320: error: too many arguments to function libpri_copy_string q931.c:3321: error: too many arguments to function libpri_copy_string q931.c:3322: error: too many arguments to function libpri_copy_string q931.c:3323: error: too many arguments to function libpri_copy_string q931.c:3324: error: too many arguments to function libpri_copy_string q931.c:3351: error: too many arguments to function libpri_copy_string q931.c:3381: error: too many arguments to function libpri_copy_string q931.c:3395: error: too many arguments to function libpri_copy_string q931.c:3396: error: too many arguments to function libpri_copy_string q931.c:3482: error: too many arguments to function libpri_copy_string q931.c:3510: error: too many arguments to function libpri_copy_string q931.c:3545: error: too many arguments to function libpri_copy_string q931.c:3573: error: too many arguments to function libpri_copy_string q931.c:3599: error: too many arguments to function libpri_copy_string q931.c:3607: error: too many arguments to function libpri_copy_string q931.c:3608: error: too many arguments to function libpri_copy_string q931.c: In function pri_internal_clear: q931.c:3695: error: too many arguments to function libpri_copy_string make: *** [q931.o] Error 1 Please help me out. Thanking you, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
