[Asterisk-Users] Error on AAHome Beta 4
Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on AAHome Beta 4
They are chown to asterisk:asterisk and chmod 777 . But I am still getting those error. Any other suggestion? Thanks Quoting asterisk [EMAIL PROTECTED]: Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 ___ Try chowning those files to asterisk. I also think there is a script that changes file ownership in the /var/aah_build directory (i am guessing here) Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk at home and Asterisk 1.2 beta
Any chance anybody has asterisk at home with asterisk 1.2 beta? any problem if I reinstall the beta on top of asterisk at home? Thanks CM ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 and IAX2
Anybody here using iax2 for one call leg and other call leg for oh323? I am getting broken sounds from Iax2 call get. Can somebody here help? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 call leg and IAX call leg
Hi, I am having a strange problem. When ever I made a call, one leg is IAX and other leg is OH323. The call establish fine but anybody talking from OH323 leg side, I hear broken sound in IAX side. Something is wrong with RTP. Is it something do with FRAME set in OH323? if so, what will be the correct set? I am using Codec 729 for both call legs. Anybody has any idea on this issue? Thanks CM Rahman Jr. CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)
I like to test it as well. Thanks Quoting Derek Whitten [EMAIL PROTECTED]: Hi Tim, I would like to test it as well. Thanks, Derek On Wed, 2005-08-03 at 00:37, Boris Zolotarev - Pamet wrote: Hello Tim, I am definitely interested in testing it. Please contact me off the list. Best Regards, Boris. If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR Server
does anybody know any CDR server in public domain or low cost? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Areski Calling Card Solution plz
I am using it. I liked it. The guy did a good job. He doesn't have the agent module yet. But I think that is on its way. Thanks Quoting Arnd Vehling [EMAIL PROTECTED]: Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 version 0.6.6.
Hi, I have downloaded asterisk-oh323-0.6.6.tar pwlib-Janus_patch4-src-tar openh323-Janus_patch4-src-tar pwlib and openh323 compiled fine as instructed. When I tried to compile asterisk-oh323 I am getting this and anybody know howto fix this? [EMAIL PROTECTED] oh323]# cd asterisk-oh323-0.6.6 [EMAIL PROTECTED] asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpmTESTS BUGS COPYINGREADMErules.mak wrapper [EMAIL PROTECTED] asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' ./check_ver /root/oh323/pwlib pwlib ./check_ver /root/oh323/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/asterisk-driver' gcc -Wall -DUSE_OLD_CAPABILITIES_API=1 -march=i686 -pipe -Wstrict-prototypes - Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -fPIC - I/usr/include/asterisk -I../wrapper -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/string.h:33, from chan_oh323.c:34: /usr/local/lib/gcc-lib/i686-pc-linux-gnu/3.3.6/include/stddef.h:213: error: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/string.h:38: error: syntax error before extern /usr/include/string.h:39: error: syntax error before __THROW /usr/include/string.h:43: error: syntax error before __THROW /usr/include/string.h:56: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:58: error: syntax error before extern /usr/include/string.h:58: error: syntax error before __THROW /usr/include/string.h:62: error: syntax error before __THROW /usr/include/string.h:66: error: syntax error before __THROW /usr/include/string.h:80: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:82: error: syntax error before extern /usr/include/string.h:83: error: syntax error before __THROW /usr/include/string.h:86: error: syntax error before __THROW /usr/include/string.h:90: error: syntax error before __THROW /usr/include/string.h:93: error: syntax error before __THROW /usr/include/string.h:97: error: syntax error before __THROW /usr/include/string.h:100: error: syntax error before __THROW /usr/include/string.h:104: error: syntax error before __THROW /usr/include/string.h:107: error: syntax error before __THROW /usr/include/string.h:160: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:162: error: syntax error before extern /usr/include/string.h:162: error: syntax error before __THROW /usr/include/string.h:164: error: syntax error before __THROW /usr/include/string.h:173: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:176: error: syntax error before extern /usr/include/string.h:177: error: syntax error before __THROW /usr/include/string.h:181: error: syntax error before __THROW /usr/include/string.h:184: error: syntax error before __THROW /usr/include/string.h:187: error: syntax error before __THROW /usr/include/string.h:192: error: syntax error before __THROW /usr/include/string.h:197: error: syntax error before extern /usr/include/string.h:199: error: syntax error before __THROW /usr/include/string.h:230: error: syntax error before extern ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show
I will need toll free for USA. If whle north america available, i would be interested as well. the incoming call will come via SIP. Thanks Quoting Leon Sun [EMAIL PROTECTED]: 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and 2 line MGCP phone
HI, Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and 2 line MGCP phone
Are you using, putting those lines in the mgcp.conf file, should handle two lines? Did anybody tried it? Thanks Quoting Florian Overkamp [EMAIL PROTECTED]: Hi, -Original Message- Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Theoretically you could differentiate by the line: aaln/[EMAIL PROTECTED] aaln/[EMAIL PROTECTED] Are typical indications for this. I've never seen a phone that does this, though.. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] matching sip connection under sip.conf
Anybody know how to match under sip.conf and cisco 53xx ? It looks like due to dynamic port number, it is not able to authorize it. Here is what I get under debug Using latest request as basis request Sending to 216.236.160.15 : 5060 (non-NAT) Found no matching peer or user for '216.236.160.15:53182' It looks like the port number is changing and that is why the * can not recognize it. Is there way to get around to this? My sip.conf [216.236.160.15] type=friend username=696 fromuser=696 host=216.236.160.15 context=from-pstn Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 \ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dynamic port problem !!
Is there a way to fix this problem? I am using cisco 5300 to connect to asterisk but it is failing to recognize by IP due to port number changing. Unfortunately, Cisco send sip request with changing port number. What can be done? No matter what I put on sip.conf, I couldn't get it to match. Any help? Here is the debug Found no matching peer or user for '216.236.160.15:50896' Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G723.1 and G729 on Athlon 64
You need to do this on intel chipset. You can not do it on AMD. I guess digium has it. Thanks Quoting Ronald Wiplinger [EMAIL PROTECTED]: I would like to install G723.1 and G729 on an Athlon 64. I looked at http://readytechnology.co.uk but I could not get a clue how to compile / get all the things for an Athlon. It seems it is only for Intel architecture, ... Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN-5300-asterisk(sip)
I know there was a posting regarding how to configure 5300 and asterisk so I can dial pstn and get connected to asterisk. Can somebody share the sip.conf and dial-peer config with me? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 5300 to asterisk
Hi, Anybody here configured 53xx to connect to asterisk ? So, if the pstn call is made it will go through autoattendant on asterisk? If you have it, please share your sip.conf and dial-peer of cisco. Thanks ** C.M. Rahman Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callback broken?
HI, Can somebody tell me how to get callback working? I have put a script in /var/spool/asterisk/outgoing but nothing happens. even the debug shows nothing is happening. But the file gets erase. How to check what is happening? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] callback broken?
I am running Asterisk @ home. Anybody know if it is turned off? My format is fine as far as I know. If it was wrong, the debug would catch it. Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Monday, April 18, 2005 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] callback broken? Check your call file format first, before assumimg that it is 'broken'. It is not broken for anyone as is evident from no complaints so far. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CM Rahman Jr. Sent: Monday, April 18, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] callback broken? HI, Can somebody tell me how to get callback working? I have put a script in /var/spool/asterisk/outgoing but nothing happens. even the debug shows nothing is happening. But the file gets erase. How to check what is happening? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPP g723 and getting error when starting asterisk
The compilation of codec g723.1 was fine. After I have copied to /usr/lib/asterisk/modules and started the asterisk -c .. I get this below error before asterisk quit. Anybody had any idea on Intel codec 723.1 ? [codec_g723.so] = (G723.1/PCM16 (signed linear) Codec Translator, based on IPP) Illegal instruction [EMAIL PROTECTED] G723.1]# Ouch ... error while writing audio data: : Broken pipe Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of denon Sent: Sunday, April 17, 2005 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IPP g729 x86_64 I'm curious, how are you licensing your codec? The source is open, but the codec usage licensing is not. I think you'll find that licensing it from Digium will be much simpler, not to mention their code will Just Work(tm) without any messing around. -d At 12:08 PM 4/17/2005, you wrote: Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 on @homeasterisk
Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks ** C.M. Rahman Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with simple callback application from newbie
Are you asking or saying it can be done that way? ** C.M. Rahman Jr. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Thursday, April 07, 2005 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help with simple callback application from newbie On Apr 6, 2005, at 8:31 PM, CM Rahman Jr. wrote: I am looking for same type of solution. Anybody here can help? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herbert Chan Sent: Wednesday, April 06, 2005 12:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help with simple callback application from newbie Hi there, I know that this has been covered many times with already but I still can't seem to get it to work. Basically, all I want to do is: 1) call the asterisk server from an external line 2) Say punch in a particular extension number 3) Asterisk then hangs up and calls me back based on my Caller ID or a pre-determined number 4) I'm then provided with a dialtone to make outgoing calls Could you have an extension that prompts for a password, then proceeds to (h)angup and begin a new DIAL which then drops you into DISA? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with simple callback application from newbie
I am looking for same type of solution. Anybody here can help? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herbert Chan Sent: Wednesday, April 06, 2005 12:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help with simple callback application from newbie Hi there, I know that this has been covered many times with already but I still can't seem to get it to work. Basically, all I want to do is: 1) call the asterisk server from an external line 2) Say punch in a particular extension number 3) Asterisk then hangs up and calls me back based on my Caller ID or a pre-determined number 4) I'm then provided with a dialtone to make outgoing calls I wanna do this cos I'm a cheapo dude with a free-incoming call plan on my mobile. I've tried and failed at using app_qcall, Its not included in the latest CVS download.. But I chucked the files in and re-installed asterisk. No directory qcall exists in /var/spool/asterisk... And I don't know if I should create one and what file should I create within that folder? Other solutions I've searched for show scripts for CAPI call back, etc.. But I just wanna callback on a zap channel. Help? Is there like an easy way to do this? Im a layman who never touched linux till a week ago.. By the was, I'm using a TDM22B with 2 anolog phones attached to 2 outgoing lines Asterisk ROCKS!!! Herbert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRV and Asterisk
HI, Anybody here used SRV feature in Asterisk to route sip calls? If anybody were able to get this working, I like to know how the sip configuration was set. I am testing this in a lab with two different domain. I was hoping Asterisk will be able to route the call by doing DNS look up on SRV. My lab test if failing. Let me know if anybody here did it. Thanks ** C.M. Rahman Jr. CCNP, MCSE Security Secure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
