Re: [asterisk-users] How to reuse mysql connection between AGI's
Hi Faheem, You need to build some daemonized application, here FastAGI will help you Regards On Fri, Aug 6, 2010 at 10:54 AM, Faheem faheem_...@yahoo.com wrote: Hey, Is there any way to share MySQL connection between different agi's. Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new MySQL connection and then execute query and then disconnects. So, Ideally there should be one connection, and it should be reused between each agi and when a call is over it should be disconnected. Is there any mechanism to reuse single MySQL connection between agi scripts? The agi scripts are written in Perl Thanks, Faheem, M. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Security - What inbound variables can attackers populate or use when calling?
I am setting filters, etc. on variables that attackers can send asterisk when they call (for example when they initially call into asterisk). So far, I am filtering: exten CALLERID(name) CALLERID(num) What other fields or variables would an attacker be able to use in the packets that they send when placing the call to asterisk? Further, I am assuming that in the case that an attacker, first, simply dials in normally and then after reaching voice prompts or other, starts his/her attack, then all I need to filter in that case is exten. Anything else here as well? Thanks!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rolling over Master.csv CDR File
Have a look into the command logrotate http://www.cyberciti.biz/faq/how-do-i-rotate-log-files/ On 05/08/10 21:26, Ujjval Karihaloo wrote: Is there a setting to roll over the Master.csv CDR File in /var/log/asterisk/cdr-csv, from and ZIP the older file once its gets a certain size? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
Steve- On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. so there is still a place for LPC10 [...] I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
Please can anyone help me with this ?! I have tried renaming the sip.conf file, or tried including another file into sip.conf like sippy.conf and then add sippy.conf = mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working. The only thing that changes something is my example listed below, but then I always have that nasty WARNING which I find odd. I need realtime sip registrations (so without having to do a sip reload). Kind regards, Jonas. On 08/03/2010 10:13 AM, Jonas Kellens wrote: Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:passw...@sip.provider.net In ext_config (text file) I have : sipusers = mysql,AsteriskDB,sip_buddies sippeers = mysql,AsteriskDB,sip_buddies sip.conf = mysql,AsteriskDB,ast_config In sip.conf (text file) I have also : sip.conf : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) After a reload I noticed that the registration came through when I executed sip show registrations. This realtime works. But I then get a lot of the following messages : / [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2'/ rtcachefriends is turned on (see above) qualify is on on every peer (and I want it to stay that way) Can anyone tell me what I need to configure to get a 100% working example ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 without DAHDI
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote: Kevin P. Fleming wrote: On 08/05/2010 03:52 PM, Roderick A. Anderson wrote: I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 installed from the asterisk.org and digium.com repositories. I have Asterisk starting (service asterisk start) but see errors about dahdi in /var/log/asterisk/messages. ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory Linux-Vservers don't allow, under normal circumstances, guests to fiddle with /dev. I could create all the entries in /dev/dahdi but as far as I can determine I have no need of dahdi -- Asterisk 1.6.2, SIP only connections, and currently no conference call needs. Is there a way to stop Asterisk (safe_asterisk) from even trying to load dahdi? Yes; don't load codec_dahdi.so in Asterisk. Use 'noload' in your modules.conf file. What packages have you installed from the asterisk.org and digium.com yum repositories? Thanks Kevin. I made that entry and now there are no more dahdi errors in the log file. Here is the command I used to install Asterisk. yum install asterisk16 asterisk16-configs asterisk16-voicemail And here are some RPM queries # rpm -qa | grep asterisk asterisk-sounds-core-en-gsm-1.4.19-1_centos5 asterisk16-core-1.6.2.10-1_centos5 asterisk16-configs-1.6.2.10-1_centos5 asterisk16-dahdi-1.6.2.10-1_centos5 asterisk16-voicemail-1.6.2.10-1_centos5 asterisk16-doc-1.6.2.10-1_centos5 asterisk16-1.6.2.10-1_centos5 If you aren't going to use DAHDI, don't install the asterisk16-dahdi package :-) Without that package, codec_dahdi wouldn't even be on your system, and you would never have had this problem. With that package (which provides codec_dahdi, among other things), the assumption is that you want that module loaded. There are also these packages. # rpm -qa | grep dahdi kmod-dahdi-linux-2.3.0.1-1_centos5.2.6.18_194.8.1.el5 dahdi-firmware-tc400m-MR6.12-1_centos5 dahdi-firmware-oct6114-128-1.05.01-1_centos5 dahdi-firmware-2.0.2-1_centos5 asterisk16-dahdi-1.6.2.10-1_centos5 kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_centos5.2.6.18_194.8.1.el5 dahdi-firmware-oct6114-064-1.05.01-1_centos5 dahdi-firmware-hx8-2.06-1_centos5 dahdi-linux-2.3.0.1-1_centos5 You don't need to have any of these DAHDI packages installed at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error chan_sip.c: sip_call: No audio format found to offer. # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on core show translation recalc 10. Can anybody please tell if missing some step in this. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
On Fri, Aug 6, 2010 at 5:29 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * Hi Go For asterisk top directory. And follow below steps to check whether speex function module is enable or not. ./configure make menuselect = Go for Dialplan Function = Then func_speex. if func_speex shows [XXX] this symbol that means func_speex module is not enable. And if you select func_speex then it shows dependency below of module list. -- Regards, Chandrakant Solanki -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote: snip MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. so there is still a place for LPC10 [...] I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. -Jeff I wonder where David Rowe's newer CODEC2 fits into this discussion? (http://codec2.org/) Clearly it's not implemented anywhere yet, but it may prove yet useful in very bandwidth constrained applications. Oh yes. It's completely open source and should not be subject to patent issues. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On Fri, 06 Aug 2010 07:40:44 -0500, Michael Graves wrote: On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote: snip MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. so there is still a place for LPC10 [...] I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. -Jeff I wonder where David Rowe's newer CODEC2 fits into this discussion? (http://codec2.org/) Clearly it's not implemented anywhere yet, but it may prove yet useful in very bandwidth constrained applications. Oh yes. It's completely open source and should not be subject to patent issues. Michael The more appropriate link should have been http://www.rowetel.com/blog/?page_id=452 Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
Hi, May you also need to install *speex-tools* . if problem retain then let us know about your Linux distribution and Asterisk version. Regards On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200
To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax messages from all ATA's to the PSTN (so ATA - Asterisk - PGW - PSTN) failed with a variety of error messages so i tested the different steps one by one. ATA's - Asterisk ReceiveFax: So far i have only succeeded in sending fax messages from the Fritz!Box 7170 to both Asterisk configurations using the ReceiveFax application. Sending fax messages from the other ATA's to Asterisk using the ReceiveFax application failed. ATA's - PGW: To exclude Asterisk i have connected the ATA's directly to the PGW; no success either. Asterisk - PGW: To exclude the ATA's i used the Asterisk SendFax application to send a TIFF file to a landline each time with a different fax machine connected to it. Results: Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 : asterisk[1367]: WARNING[18591]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=19: Received other than DIS while waiting for DIS. asterisk[1367]: WARNING[18591]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18906]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=20: Received no response to DCS or TCF. asterisk[1367]: WARNING[18906]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18986]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. asterisk[1367]: WARNING[18986]: app_fax.c:817 in transmit: Transmission error Asterisk 1.6.2.6 with Fax for Asterisk : asterisk[7092]: WARNING[3167]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f49' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3167]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f49' in T.38 mode asterisk[7092]: VERBOSE[3226]: -- FAX handle 0: [ 028.000627 ], entering CLOSING state asterisk[7092]: VERBOSE[3225]: -- Channel 'SIP/out.to.pgw-000b3f72' FAX session '11' is complete, result: 'FAILED' (FAX_FAILURE_TRAINING), error: '3RD_FRM_CHECK_ERROR', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' asterisk[7092]: WARNING[3272]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f8b' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3272]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f8b' in T.38 mode My questions: - Does anyone have experience with T.38 fax with a setup like this: ATA - Asterisk - PGW - PSTN? - Does anyone have experience in connecting Asterisk to a Cisco PGW 2200 + AS5400XM? - Are there any tools to debug T.38 traffic? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR report
2010/8/5 Dario Quiroz darioqui...@gmail.com Hi all! Are someone using a CDR report? I have an Asterisk 1.6 running perfect but I need a web based report of CDRs. Nothing big, only the basic. Have anybody a how-to or a link? Look for asterisk-stats (or cdr-stats as it has been renamed) : users are rather pleased with it. Thanks in advance!! -- Atenciosamente, --- Dario Quiroz Analista de Suporte (71) 9275-9080 darioqui...@gmail.com --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk-Users] How do I install speex for
Hi Chandrakant I have checked and it shows func_speex module is enabled. Where can I install speex-tools from ? Asterisk version 1.6.2.10 and Centos 5.5 are installed. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using a 1.4 config with 1.6
I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. TIA, Rod -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a 1.4 config with 1.6
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A. Anderson Subject: [asterisk-users] Using a 1.4 config with 1.6 I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? TIA, Rod I am doing dual testing with 1.4.3X and 1.6X and the only thing that has hit me so far is having to change the dialplan delimiters (from | to ,) If your dialplan is simple enough, this may not affect you at all. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does deny/permit work in sip.conf?
I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] How do I install speex for
Hi, Currently CentOS yum repository does not provide speex-tools so you have to install it manaully. follow the steps given below. 1. first remove existing speex pacages yum remove speex* 2. run following to install required rpms rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-1.2-0.10.rc1.i386.rpm rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-devel-1.2-0.10.rc1.i386.rpm rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-tools-1.2-0.10.rc1.i386.rpm 3. reconfigure asterisk 4. verify speex codec via make menuselect under Codec Translators. if speex is enabled and selected continue with make Enjoy! On Fri, Aug 6, 2010 at 7:13 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi Chandrakant I have checked and it shows func_speex module is enabled. Where can I install speex-tools from ? Asterisk version 1.6.2.10 and Centos 5.5 are installed. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a 1.4 config with 1.6
Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A. Anderson Subject: [asterisk-users] Using a 1.4 config with 1.6 I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? TIA, Rod I am doing dual testing with 1.4.3X and 1.6X and the only thing that has hit me so far is having to change the dialplan delimiters (from | to ,) If your dialplan is simple enough, this may not affect you at all. Excellent! Mine is very simple. Open/Closed hours, four menu options. Thanks, Rod -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a 1.4 config with 1.6
On Fri, Aug 6, 2010 at 10:36 AM, Roderick A. Anderson raand...@cyber-office.net wrote: I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. Be sure to read CHANGES and UPGRADE.txt within the source directory of Asterisk. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Grandstream GXV3140
Hi, Do any of you have these phones ? How have you found it ? Are you using them over WiFi or hard wired ? Does it play nicely with Asterisk ? Need to replace my Snom M3s and this phone maybe a contender. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip address or host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and TE420P
I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and TE420P
On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.comwrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and TE420P
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Subject: Re: [asterisk-users] Asterisk 1.4 and TE420P On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.com wrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. To expand on OP's question, is he going to have to upgrade to 1.4.3X/DAHDI to make the TE420P work? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. On Fri, 2010-08-06 at 10:57 +0200, Jonas Kellens wrote: Please can anyone help me with this ?! I have tried renaming the sip.conf file, or tried including another file into sip.conf like sippy.conf and then add sippy.conf = mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working. The only thing that changes something is my example listed below, but then I always have that nasty WARNING which I find odd. I need realtime sip registrations (so without having to do a sip reload). Kind regards, Jonas. On 08/03/2010 10:13 AM, Jonas Kellens wrote: Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:passw...@sip.provider.net In ext_config (text file) I have : sipusers = mysql,AsteriskDB,sip_buddies sippeers = mysql,AsteriskDB,sip_buddies sip.conf = mysql,AsteriskDB,ast_config In sip.conf (text file) I have also : sip.conf : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) After a reload I noticed that the registration came through when I executed sip show registrations. This realtime works. But I then get a lot of the following messages : [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' rtcachefriends is turned on (see above) qualify is on on every peer (and I want it to stay that way) Can anyone tell me what I need to configure to get a 100% working example ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and TE420P
On Fri, Aug 6, 2010 at 11:24 AM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Subject:* Re: [asterisk-users] Asterisk 1.4 and TE420P On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.com wrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. To expand on OP’s question, is he going to have to upgrade to 1.4.3X/DAHDI to make the TE420P work? I've only ever used it with DAHDI. I've used it with version of asterisk down to 1.4.22 though, I'm pretty sure. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On 08/06/2010 04:43 PM, Jeff Brower wrote: Steve- On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. I think all the IP for MELP is now in the hands of Compandent, and TI no longer has the ability to waive royalties. Either way, government use and use with TI silicon are two niches that might work out well, and everything else is a problem for several more years. If you are going to pay royalties for a low bit rate codec, IMBE is probably a better option. TI is a good option, but what do you have against Mindspeed? Choosing a good option for this kind of card is mostly about managing the patent licence fees. I assume Mindspeed gave Digium the best option for doing that, within Digium's volume constraints. so there is still a place for LPC10 [...] I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. MELPe is definitely a compandent thing, and TI cannot waive fees for that. MELP and MELPe are derived from LPC10. Any attempt to improve LPC10 would take you down a similar road, though you would need to skirt around the patents. Do you really consider MELPe to be an enormous improvement over LPC10? Its still pretty lousy compared to a number of options at about 5kbps, and RTP overheads mean the gain from going lower than 5k isn't that big. The main reason LPC10 and MELPe offer a low bit rate in RTP is the minimum packet you can pack 22.5ms frames into sanely is a 90ms one. 90ms RTP *really* cuts the overheads, compared to the more typical 20ms or 30ms packets used for G.729. As others have mentioned, David Rowe is working on a modern 2400bps codec. He did a burst of work some time ago, and then put it aside while busy with other things. He recently told me he is restarting the work, and he wants to get that codec into good shape before the end of this year. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Grandstream GXV3140
- Original Message - On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote: Hi, Do any of you have these phones ? How have you found it ? Are you using them over WiFi or hard wired ? Does it play nicely with Asterisk ? Need to replace my Snom M3s and this phone maybe a contender. Full of bugs. Stay away. Vendors have started dropping this phone. I use them personally when traveling (to talk to my baby girl), but as a commercial product this phone is AWFUL. j Oh, that is very disappointing indeed; especially some of the others bits I read on the net :( Desperately trying to find a new phone that will support: * Centralised management and deployment * OpenVPN * Asterisk support The company I work for have got very despondent with Snom and their lack of support. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and OCS2007 R2
Hi all, i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for OCS2007 R2 following the HowTo http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx. I can call the OCS from Asterisk and vice versa. The only thing that doesn't work is displaying the callerid(name) in OCS Communicator 2007. I can manipulate the callerid(num) in Asterisk to any value I want using Set(), but callerid(name) is not displayed. Has anyone a similar setup with names being displayed in OCS? Thanks in advance. Oliver -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reinstalling Asterisk due to hardware changes
My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to put into my Asterisk box. I know from the wonderful O'Reilly book that the proper installation is Zaptel à libpri à Asterisk. Is it possible to simply reinstall in that order once I have installed the card and have Asterisk successfully work or is it best to uninstall all three packages and install from scratch? Thanks, Jeremy Jeremy Hellstrom Specialist, IT/Systems/Sampling Phone (604) 664 2472 Fax (604) 664 2456 Suite 1550 1090 West Georgia St. Vancouver BC V6E 3V7 www.synovate.com http://www.synovate.com/ Vancouver's ultimate focus group facility State-of-the-art focus group facility, now with FocusVision! Synovate is your one-stop shop for all qualitative research needs from recruiting, to moderating, to facilitating. To learn more, visit www.synovate.com/vancouverfocusgroups http://www.synovate.com/vancouverfocusgroups -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reinstalling Asterisk due to hardware changes
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jeremy.hellst...@synovate.com Subject: [asterisk-users] Reinstalling Asterisk due to hardware changes My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to put into my Asterisk box. I know from the wonderful O'Reilly book that the proper installation is Zaptel -- libpri -- Asterisk. Is it possible to simply reinstall in that order once I have installed the card and have Asterisk successfully work or is it best to uninstall all three packages and install from scratch? In my experience, all you should need to do is install Zaptel/Dahdi and libpri, then do a new make make install on your existing installation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. I think all the IP for MELP is now in the hands of Compandent, and TI no longer has the ability to waive royalties. That is not correct. Compandent has filed copyrights on certain files associated with a C549 chip assembly language implementation they did under contract to NSA around 2001. TI has patent rights on 2400 bps, TI + Microsoft on 1200 bps, and TI + Microsoft + Thales Group on 600 bps. Microsoft's IP came about as a result of acquiring a company called SignalCom around 2001. If the noise pre-processor is used, then there is some ATT IP. To verify this, you can search dsprelated.com (specifically, look for posts discussing this issue on comp.dsp), and you can also read the Compandent IPR section of the MELPe Wikipedia page (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction). That section was authored by the Compandent's founder, Oded Gottesman. Oded is a super sharp, very hard working guy. Compandent also claims a copyright on some C code in the file melp_syn.c (synthesis filter). I have read discussions by DSP experts indicating the copyrighted section of code can be implemented in alternative ways, but Oded may say that's not accurate. Either way, government use and use with TI silicon are two niches that might work out well, and everything else is a problem for several more years. If you are going to pay royalties for a low bit rate codec, IMBE is probably a better option. I would disagree because IMBE source is not available. MELPe source is available and can be downloaded online. TI is a good option, but what do you have against Mindspeed? Choosing a good option for this kind of card is mostly about managing the patent licence fees. I assume Mindspeed gave Digium the best option for doing that, within Digium's volume constraints. My understanding in talking to Digium engineers at Globalcom and other trade shows back in 2006 is they were worried about interfacing the TI TNET series devices over the PCI bus. They would have needed an FPGA with some non-trivial logic programming, so I understand their decision. But if they had got past their FPGA writer's block, they could have put one TNETV3010 chip on there, even smaller than the Mindspeed and without the heat sink, and had twice the channel capacity as they do now. so there is still a place for LPC10 [...] e I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. MELPe is definitely a compandent thing, and TI cannot waive fees for that. MELP and MELPe are derived from LPC10. Any attempt to improve LPC10 would take you down a similar road, though you would need to skirt around the patents. Again, not correct. Suggest to research the many online independent sources, or contact NSA (who initiated the overall MELPe effort in the 1990s, in response to a need to significantly improve over LPC10) and who can give you a complete IP list. Do you really consider MELPe to be an enormous improvement over LPC10? Its still pretty lousy compared to a number of options at about 5kbps, and RTP overheads mean the gain from going lower than 5k isn't that big. The main reason LPC10 and MELPe offer a low bit rate in RTP is the minimum packet you can pack 22.5ms frames into sanely is a 90ms one. In MOS terms, yes. In VoIP terms, I agree it's not clear cut. At 2400 bps, a 90 msec packet would be 27 payload bytes. For IP/UDP/RTP usage, that much delay could well be counterproductive. Places where I have seen MELPe effectively used for VoIP include applications
[asterisk-users] Asterisk and OCS2007 R2
Hi all, i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for OCS2007 R2 following the HowTo http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx. I can call the OCS from Asterisk and vice versa. The only thing that doesn't work is displaying the callerid(name) in OCS Communicator 2007. I can manipulate the callerid(num) in Asterisk to any value I want using Set(), but callerid(name) is not displayed. Has anyone a similar setup with names being displayed in OCS? Thanks in advance. Oliver -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OCS2007 R2
Oliver, Did you happen to use a Dialog Media gateway in the mix, or is this straight Asterisk to OCS? We are implementing this in my shop, and running a bit of a ground (trying to use the MediaGateway). Any help anyone could provide would be wonderful. Thanks all. Lyle J. McKarns --- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of unsero...@aol.com Sent: Friday, August 06, 2010 3:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS2007 R2 Hi all, i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for OCS2007 R2 following the HowTo http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx. I can call the OCS from Asterisk and vice versa. The only thing that doesn't work is displaying the callerid(name) in OCS Communicator 2007. I can manipulate the callerid(num) in Asterisk to any value I want using Set(), but callerid(name) is not displayed. Has anyone a similar setup with names being displayed in OCS? Thanks in advance. Oliver -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OCS2007 R2
I did it straight Asterisk to OCS using the OCS Mediation Server. We do have Dialogic Diva Server Cards which are able to be used as Media Gateway too using an additional software called SipControl (not 100% sure about the name) but as this software needs to be licensed separately I prefer a direct setup like a actually have. Oliver Oliver, Did you happen to use a Dialog Media gateway in the mix, or is this straight Asterisk to OCS? We are implementing this in my shop, and running a bit of a ground (trying to use the MediaGateway). Any help anyone could provide would be wonderful. Thanks all. Lyle J. McKarns --- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of unsero...@aol.com Sent: Friday, August 06, 2010 3:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS2007 R2 Hi all, i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for OCS2007 R2 following the HowTo http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx. I can call the OCS from Asterisk and vice versa. The only thing that doesn't work is displaying the callerid(name) in OCS Communicator 2007. I can manipulate the callerid(num) in Asterisk to any value I want using Set(), but callerid(name) is not displayed. Has anyone a similar setup with names being displayed in OCS? Thanks in advance. Oliver = -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip address or host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- Does permit/deny have any influence on registration, or is it related to the destinations it can call to or receive call from? How do you stop an asterisk server from accepting registrations when the IP is outside a subnet even if the username and secret are correct? When host=dynamic registrations are accepted even if the pemit IP is different from the registered device's IP address. Does permit/deny work on a single IP address eg 192.168.4.111/255.255.255.2555 The same seems to apply in the [general] section, with contactdeny and contacnt permit When I set contactdeny=0.0.0.0/0.0.0.0 contactpermit=192.168.4.111/255.255.255.255 Devices whose IP is not 192.168.4.111 are able to register. _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need USA DIDs
Sorry for asking this after 6 week of staleness on this post, but do you know which FCC registration is needed? We already have a 214. Is that it? Thanks. Regards HASSAN On Thu, Jun 24, 2010 at 00:43, Tarek Sawah tareksa...@hotmail.com wrote: i consuleted didforsale.com regarding the wholesale thing and their response was that you should buy a bulk of numbers and make your own api.. one more thing.. if you are in the USA ..be sure to start your FCC registration (if you don't have it yet) because it can be a disaster for US companies providing DID numbers to US citizens without FCC license. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 -- Date: Wed, 23 Jun 2010 23:43:14 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Need USA DIDs On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick r...@readywire.com wrote: Agreed! Didforsale.com is THE way to go. -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Anyone having experience with didww.com ? Sorry, I forgot to mention I am looking for wholesale DID -- reseller option with API to that my customers can select country - city -- DID from my website. Thx -- The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. Get busy.http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reuse mysql connection between AGI's
On Thu, 5 Aug 2010, Faheem wrote: Hey, Is there any way to share MySQL connection between different agi's. No. Each AGI is executed as a separate process. While debugging (not interacting with a real call), you can executed your AGI completely independent from Asterisk from the command line if you feed it appropriate cruft via stdin. Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new MySQL connection and then execute query and then disconnects. So, Ideally there should be one connection, and it should be reused between each agi and when a call is over it should be disconnected. Is there any mechanism to reuse single MySQL connection between agi scripts? The agi scripts are written in Perl I suspect the issue is Perl, not MySQL. If you are doing something often enough to be a performance issue, why did you write it in an interpreted script language instead of a compiled language? In previous experiments, I demonstrated you can execute XXX AGIs written in c in the time you can load the interpreter, parse your script, and execute a single AGI written in Perl or PHP. In that experiment, the null-agi only read the AGI environment and exited. I added mysql_init(), mysql_real_connect(), and mysql_close() and... On my wimpy (by current standards) 500MHz AMD Geode I can execute about 20 null-agi's per second. On my almost as wimpy 1.1GHz AMD 8650 I can execute about 100 per second. I'd suggest re-implementing your AGIs in c and merging a couple of the AGIs together if it makes sense for your environment. As an alternative, you could go the fastagi() route. This means changing your code to execute as a daemon which means you can keep the same MySQL connection and you eliminate the AGI process creation overhead. It also introduces complexity in handling simultaneous call execution, you become dependent on that process being available, and you lose some flexibility in being able to make changes to your AGIs without affecting calls in progress. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
On 08/06/2010 06:45 PM, Carlos Chavez wrote: You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. Using the EXEC-command in sip.conf means I will have to issue a sip reload when I want to load changes in the database ?! New information that is put into the REGISTER-database is not available without a 'sip reload' ?! If not, do you see another way to have new registrations without a 'sip reload' ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
On 08/06/2010 06:45 PM, Carlos Chavez wrote: Or use the realtime static table for everything. What do you mean by everything ?! What is this everything ?! You mean all the sip options in a database and so no sip.conf file ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. Using the EXEC-command in sip.conf means I will have to issue a sip reload when I want to load changes in the database ?! New information that is put into the REGISTER-database is not available without a 'sip reload' ?! If not, do you see another way to have new registrations without a 'sip reload' ?! Kind regards, Jonas. -- Why don't you use 'real' realtime meaning to have your sip peers in your database? Then you would not have to do a reload after adding new peers to your db. And you can still have sip peers additionally in sip.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
This works. I have tested with the following settings: In regards to the specifics of your question: In sip.conf: dynamic_exclude_static=yes In users.conf, for each user (changing the permit statement to the ip of each user): hassip=yes host=dynamic registersip=yes deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 (using your ip setting) Hope that helps -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church Sent: Friday, August 06, 2010 11:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How does deny/permit work in sip.conf? I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set outgoing number in filename of the recordings
Hi,
Im not able to set the outgoing number in filename for asterisk recordings
Following is what I have done in
/var/lib/asterisk/agi-bin/recordingcheck file
.
.
.
include(phpagi.php);
/**/
$agi = new AGI();
$temp = $agi-get_variable(agi_dnid) ; // I have also tried with
get_variable(DIAL_NUMBER)
if($temp['result'] == 1 ) {
$dnid = $temp['data'] ;
}
else
{
$dnid = NUMBER ;
}
$timestamp = $argv[1];
$uniqueid = $argv[2];
$type = $agi-get_variable(ARG2);
.
.
.
Please help me
Regards,
Rishi
--
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Re: [asterisk-users] How does deny/permit work in sip.conf?
On 08/06/2010 02:16 PM, Frank Church wrote: On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip address or host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- Does permit/deny have any influence on registration, or is it related to the destinations it can call to or receive call from? How do you stop an asterisk server from accepting registrations when the IP is outside a subnet even if the username and secret are correct? When host=dynamic registrations are accepted even if the pemit IP is different from the registered device's IP address. Does permit/deny work on a single IP address eg 192.168.4.111/255.255.255.2555 The same seems to apply in the [general] section, with contactdeny and contacnt permit When I set contactdeny=0.0.0.0/0.0.0.0 contactpermit=192.168.4.111/255.255.255.255 Devices whose IP is not 192.168.4.111 are able to register. When I've used permit/deny, I did it in conjunction with insecure set to port,invite to allow gateways that didn't register and don't use username/secret to originate calls but only from the ip range in permit. In fact it was for a provider that had gateways on a large number of IP addresses, all in the same CIDR block and I didn't want to do an entry for each of more than 100 gateways. contactpermit/contactdeny *should* work as you are suggesting that you want I've never tried that. I may attempt it tonight and see on my 1.4 system. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On 08/07/2010 03:15 AM, Jeff Brower wrote: Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.comwrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. I think all the IP for MELP is now in the hands of Compandent, and TI no longer has the ability to waive royalties. That is not correct. Compandent has filed copyrights on certain files associated with a C549 chip assembly language implementation they did under contract to NSA around 2001. TI has patent rights on 2400 bps, TI + Microsoft on 1200 bps, and TI + Microsoft + Thales Group on 600 bps. Microsoft's IP came about as a result of acquiring a company called SignalCom around 2001. If the noise pre-processor is used, then there is some ATT IP. To verify this, you can search dsprelated.com (specifically, look for posts discussing this issue on comp.dsp), and you can also read the Compandent IPR section of the MELPe Wikipedia page (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction). That section was authored by the Compandent's founder, Oded Gottesman. Oded is a super sharp, very hard working guy. Compandent also claims a copyright on some C code in the file melp_syn.c (synthesis filter). I have read discussions by DSP experts indicating the copyrighted section of code can be implemented in alternative ways, but Oded may say that's not accurate. That guy is PITA. He must have driven a lot of people away from MELP by the way he acts. He really annoys the regulars in the comp.dsp group by posting astroturf questions about MELP, and giving astroturf replies about how fantastic it is. That probably shapes a lot of my attitude to MELP. :-) Either way, government use and use with TI silicon are two niches that might work out well, and everything else is a problem for several more years. If you are going to pay royalties for a low bit rate codec, IMBE is probably a better option. I would disagree because IMBE source is not available. MELPe source is available and can be downloaded online. Depends what you mean by available. IMBE is patented, just like MELP is patented. Licence either, and implementations are available. IMBE has the great benefit of being widely used for commercial and amateur low bit rate channels. For example, amateur radio uses IMBE - an anomaly which is one of the drivers for David Rowe's work on an open low bit rate codec. Transcoding at low bit rates is a disaster, so using a codec you won't need to transcode is a big plus. TI is a good option, but what do you have against Mindspeed? Choosing a good option for this kind of card is mostly about managing the patent licence fees. I assume Mindspeed gave Digium the best option for doing that, within Digium's volume constraints. My understanding in talking to Digium engineers at Globalcom and other trade shows back in 2006 is they were worried about interfacing the TI TNET series devices over the PCI bus. They would have needed an FPGA with some non-trivial logic programming, so I understand their decision. But if they had got past their FPGA writer's block, they could have put one TNETV3010 chip on there, even smaller than the Mindspeed and without the heat sink, and had twice the channel capacity as they do now. TI have had DSP chips with a PCI interface for years, so that explanation doesn't make a lot of sense. Of course, these days you need a PCI-E interface. I'm not so sure about the status of those in DSP chips. so there is still a place for LPC10 [...] e I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. MELPe is definitely a compandent thing, and TI cannot waive
Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?
What kind of attack can they reform calling in? On Aug 6, 2010 1:12 AM, jwex...@mail.usa.com wrote: I am setting filters, etc. on variables that attackers can send asterisk when they call (for example when they initially call into asterisk). So far, I am filtering: exten CALLERID(name) CALLERID(num) What other fields or variables would an attacker be able to use in the packets that they send when placing the call to asterisk? Further, I am assuming that in the case that an attacker, first, simply dials in normally and then after reaching voice prompts or other, starts his/her attack, then all I need to filter in that case is exten. Anything else here as well? Thanks!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
On 08/06/2010 07:30 PM, Bruce Ferrell wrote: On 08/06/2010 02:16 PM, Frank Church wrote: On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip address or host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- Does permit/deny have any influence on registration, or is it related to the destinations it can call to or receive call from? How do you stop an asterisk server from accepting registrations when the IP is outside a subnet even if the username and secret are correct? When host=dynamic registrations are accepted even if the pemit IP is different from the registered device's IP address. Does permit/deny work on a single IP address eg 192.168.4.111/255.255.255.2555 The same seems to apply in the [general] section, with contactdeny and contacnt permit When I set contactdeny=0.0.0.0/0.0.0.0 contactpermit=192.168.4.111/255.255.255.255 Devices whose IP is not 192.168.4.111 are able to register. When I've used permit/deny, I did it in conjunction with insecure set to port,invite to allow gateways that didn't register and don't use username/secret to originate calls but only from the ip range in permit. In fact it was for a provider that had gateways on a large number of IP addresses, all in the same CIDR block and I didn't want to do an entry for each of more than 100 gateways. contactpermit/contactdeny *should* work as you are suggesting that you want I've never tried that. I may attempt it tonight and see on my 1.4 system. To follow up on my own reply. I just tried this with one of my standard peers that I use for a softphone on a 1.6.2.10 and see the registration attempt come in at the console and a warning comes up : Host '192.0.2.40' disallowed by contact ACL (violating IP 192.0.2.40) : Registration denied because of contact ACL The peer does show in sip show peers and the softphone (twinkle) shows a Registration Fails with a 603 denied. So I'd say it's working -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
On 7 August 2010 03:54, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:30 PM, Bruce Ferrell wrote: On 08/06/2010 02:16 PM, Frank Church wrote: On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip address or host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- Does permit/deny have any influence on registration, or is it related to the destinations it can call to or receive call from? How do you stop an asterisk server from accepting registrations when the IP is outside a subnet even if the username and secret are correct? When host=dynamic registrations are accepted even if the pemit IP is different from the registered device's IP address. Does permit/deny work on a single IP address eg 192.168.4.111/255.255.255.2555 The same seems to apply in the [general] section, with contactdeny and contacnt permit When I set contactdeny=0.0.0.0/0.0.0.0 contactpermit=192.168.4.111/255.255.255.255 Devices whose IP is not 192.168.4.111 are able to register. When I've used permit/deny, I did it in conjunction with insecure set to port,invite to allow gateways that didn't register and don't use username/secret to originate calls but only from the ip range in permit. In fact it was for a provider that had gateways on a large number of IP addresses, all in the same CIDR block and I didn't want to do an entry for each of more than 100 gateways. contactpermit/contactdeny *should* work as you are suggesting that you want I've never tried that. I may attempt it tonight and see on my 1.4 system. To follow up on my own reply. I just tried this with one of my standard peers that I use for a softphone on a 1.6.2.10 and see the registration attempt come in at the console and a warning comes up : Host '192.0.2.40' disallowed by contact ACL (violating IP 192.0.2.40) : Registration denied because of contact ACL The peer does show in sip show peers and the softphone (twinkle) shows a Registration Fails with a 603 denied. So I'd say it's working -- I am using 1.4.27 and it doesn't seem to work. I should probably try the 1.6 series _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?
Well, I'm not sure actually. I was attacked in June by someone who racked up between $800 and $900 in international calls to places in the middle of Africa, Korea, etc. So, I am motivated to secure this. I have made it much much more secure, definitely, but am looking for as many ways to further lock this down as possible. I figure that I should filter every field that someone could possible interact with Asterisk in case they send characters that might breach security and allow them some kind of access. Symbols like the amperstand (), comma (,), forward slash (/), at (@), pipe (|), etc. I would guess could be bad. Someone from Amsterdam was trying to register yesterday using an automated program which tried roughly 1,000 or so username password combinations before I shut asterisk down and added his/her ip to iptables to drop it. I wonder if I can configure the system to automatically detect such an attack in progress (e.g., a 1,000+ registration failures from the same ip is an 'attack') and the ip's to iptables, hosts.deny, etc. on the fly. That might be another topic I guess? This experience has emphasized the importance of securing the system and security in asterisk in general. Any insight on this would be really appreciated! Thanks!! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mike mosier Sent: Saturday, August 07, 2010 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling? What kind of attack can they reform calling in? On Aug 6, 2010 1:12 AM, jwex...@mail.usa.com wrote: I am setting filters, etc. on variables that attackers can send asterisk when they call (for example when they initially call into asterisk). So far, I am filtering: exten CALLERID(name) CALLERID(num) What other fields or variables would an attacker be able to use in the packets that they send when placing the call to asterisk? Further, I am assuming that in the case that an attacker, first, simply dials in normally and then after reaching voice prompts or other, starts his/her attack, then all I need to filter in that case is exten. Anything else here as well? Thanks!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?
On Fri, Aug 6, 2010 at 10:53 PM, jwex...@mail.usa.com wrote:
Someone from Amsterdam was trying to register yesterday using an automated
program which tried roughly 1,000 or so username password combinations
before I shut asterisk down and added his/her ip to iptables to drop it. I
wonder if I can configure the system to automatically detect such an attack
in progress (e.g., a 1,000+ registration failures from the same ip is an
‘attack’) and the ip’s to iptables, hosts.deny, etc. on the fly. That might
be another topic I guess?
Use fail2ban. Also, read some of the security advisories from earlier this
year about being sure to always use a FILTER statement whenever you're
dialing using a variable (most notably ${EXTEN}).
http://downloads.asterisk.org/pub/security/AST-2010-002.html
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
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Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
Hi Can you tell me which Linux OS are you used and what is speex / speex-devel version. Can you give details for above? -- Regards, Chandrakant Solanki On Fri, Aug 6, 2010 at 6:22 PM, Nasir Iqbal na...@ictinnovations.comwrote: Hi, May you also need to install *speex-tools* . if problem retain then let us know about your Linux distribution and Asterisk version. Regards On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shrinkcallerid
Am I really the only one having problems with this new shrinkcallerid? I can't find anything on Google about it. Was happening on 1.6.2.10 and now on 1.8.0-beta2 In sip.conf shrinkcallerid=no, yet a name like Joe Smith ends up being JoeSmith Whoever though this up anyway is stupid. Why would you want to strip spaces out of a caller ID? Is there a fix? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi issue on sangoma A200
Hi All, I have Sangoma A200 Card installed on my system, I have centos 5.5 with 64 bit, Here are the description for asterisk and dahdi. Asterisk 1.6..2.9 Dahdi: 2.3.0.1 I have two issues with dahdi 1) I am not getting full callerid on my phones from sangoma card to asterisk users. if i am connecting analog phone directly then i am getting callerid properly. I am in india and using Airtel Connection, I have set variables in chan_dahdi.conf as well for callerid but the not getting full digits in callerid, it is coming with 8 digits only. 2) Another issue is when I am hanging up the phone from inbound or outbound from the dahdi channel, it takes 5-6 seconds to dropping the call. Here are the confguration file for chan_dahdi.conf - ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-07-30 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes callerid=asreceived hanguponpolarityswitch=yes answeronpolarityswitch=yes ;cidstart=ring cidstart=polarity_IN ;cidsignalling=dtmf cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no useincomingcalleridondahditransfer=yes ;callerid=asreceived ;Sangoma AFT-A200 [slot:4 bus:2 span:1] wanpipe1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 2 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 3 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 4 --- Please hemp me for this issues. Thanks, Max Alex Voip Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
