Re: [asterisk-users] End-To-End Secured Communications
Dear Kevin, Thanks for your answer. At least in this case, only TOP DOGS must be encrypted End-To-End while they are talking between them.. so Asterisk should be right solution, they would not take advantage of .. some Asterisk Features while they are talking between them, but all other extensions and when they talk with any other person, they would take full advantage of asterisk features. I've read ZRTP works this way end-to-end and man in the middle is not possible because end points negotiate security directly through RTP which is gonna flow between end points directly. But.. only softphones availables AFAIK Is possible to secure calls end-to-end with SRPT ? Thanks in advance. Best Regards, El 5/3/2012 9:22 AM, Kevin P. Fleming escribió: On 05/03/2012 07:17 AM, Fernando Berretta wrote: Hi, I'm analyzing how to make Asterisk communications secured End-To-End, and not sure which is the best approach, SRTP + TLS seems to be secured but.. at least by default, doesn't appear to be End-To-End allowing Asterisk administrators to wiretap communications.. some sites I've hear that with SRTP is also possible End Points exchange keys between them directly avoiding Man in the Middle, is it possible with asterisk ? how On the other hand I've found ZRTP seems to be secured end-to-end, but we couldn't find any IP phones with support for it.. just SoftPhones Could someone please point me to the right direction ? This is a fundamental architectural issue with all back-to-back User Agents used in SIP networks. They are pretty much by definition a 'man in the middle'. If they are used, the administrators will have access to call signaling and media for all calls passing through them. It is also important to realize that if you want end-to-end media security, then you would not be able to use any of Asterisk's features that involve media handling (transcoding, recording, whispering/spying, music-on-hold, conferencing, etc.) Given that, what you really want is a pure SIP proxy like Kamailio or OpenSIPs. -- _ -- 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] Softphone RTP Jitter
Hi, I'm having problems with audio coming to my asterisk PBX from eyebeam softphone running on Windows XP machine, it has a lot of Jitter. I've taken a lot of tcpdumps from Asterisk machine and jitter comes only from softphones, audio comming and going to gateways, IP phones, etc has no jitter. We have installed other softphones with same result, seems the problem is in windows machine, if I ping asterisk ip all packets are back 1ms so.. doesn't seems to be a network problem. I've tried also with 2 different audio cards, one pci and one plantronics usb and problem is still there, as a work around I've forced asterisk to use a JitterBuffer of 100ms and all goes well, but I would like to find out why packets from softphone are arriving with jitter. Has someone had a similar problem ? any help would be appreciated. Best Regards, Fernando -- _ -- 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] Core Dump - Asterisk 1.4.24 - Elastix
Tzafrir Cohen wrote: On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote: Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 When Asterisk is installed from a binary package, you will normally get binaries without debug symbols to save space. You sypically don't need the debug information. However for the cases you do need them, the debug symbols are available in a separate package (with rpm: the *-debuginfo packages , with debs: -dbg packages or several similar things). Debug symbols are not needed at core dumping time. They are only needed when you try to get something useful from a core dump (e.g. with gdb). So you can install them after dumping the core. However they must be of exacatly the same version of the Asterisk package installed on your system (that dumped the core in the first place). So basically: just install the package asterisk-debuginfo and try aagain. There's also a similar package for glibc for the libc stuff, though this typically is less useful. Dear Tzafrir, I was looking for asterisk-debuginfo package for the version I've installed but..I'm not being able to find it. What can I do ? Best Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix
Dear Tzafrir, Thanks for your information, I'm gonna look for this package and will try to analyze dump file after debug package installation. Best Regards, Fernando Tzafrir Cohen wrote: On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote: Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 When Asterisk is installed from a binary package, you will normally get binaries without debug symbols to save space. You sypically don't need the debug information. However for the cases you do need them, the debug symbols are available in a separate package (with rpm: the *-debuginfo packages , with debs: -dbg packages or several similar things). Debug symbols are not needed at core dumping time. They are only needed when you try to get something useful from a core dump (e.g. with gdb). So you can install them after dumping the core. However they must be of exacatly the same version of the Asterisk package installed on your system (that dumped the core in the first place). So basically: just install the package asterisk-debuginfo and try aagain. There's also a similar package for glibc for the libc stuff, though this typically is less useful. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix
Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 Any help would be appreciated. Best Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Server Fine Tuning for Best Performance
Hi, Is there some asterisk fine tunning documentation related with System Hardware Optimizations, Operating System Tuning, Network Stack Tuning, Asterisk Settings, Network Hardware Settings, etc to get the best performance possible? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
Has SalesForce a free CRM release or only commercial one ? Bob G wrote: I just finished an integartion with SalesForce.com's connector and it worked pretty good. - Original Message - From: Fernando Berretta To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk - CRM Integration Date: Tue, 29 Apr 2008 18:16:40 -0300 Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a *free e-mail *account today at www.mail.com http://www.mail.com/Product.aspx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
Thanks for the info, I'm gonna try it Michiel van Baak wrote: On 18:16, Tue 29 Apr 08, Fernando Berretta wrote: Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Have a look at Covide: http://sourceforge.net/projects/covide /shameless_plug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
Olivier, Us much as possible would be the best but.. more importants for me are: Which feature are you specifically looking for ? 1 screen popup 2 Click2Call 3 Journaling calls inside CRM campaign, leaads and contacts Do you plan to use it in a call center or casual business office ? Call Center Regards, Olivier wrote: To me, CRM-Asterisk integration has several meanings. It could refer to : - basic click2call feature from CRM contact or project panel, - journaling Asterisk incoming and outgoing calls inside CRM projects data, - programming and executing Conference calls defined inside CRM projects data - screen popup - etc... Which feature are you specifically looking for ? Do you plan to use it in a call center or casual business office ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - CRM Integration
Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - CRM Integration
Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit calls when using autodial
I think the only way is managing the number of files in the outbound directory http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Tong wrote: Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
I haven't tried all of them but.. at least Linksys ATA AdminGuide doesn't specify such limitation FAX Enable T38 To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. The default is yes Thomas Kenyon wrote: Fernando Berretta wrote: Tzafir, I'm sorry, my question wasn't clear. Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some modifications on app_fax so the questions are: 1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card and this FXO port is forwarded to other ATA/Gateway is asterisk going to transmit this fax using t38 ? Excuse my ignorance, but don't ATAs generally only support T.38 Origination? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Tzafir, I'm sorry, my question wasn't clear. Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some modifications on app_fax so the questions are: 1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card and this FXO port is forwarded to other ATA/Gateway is asterisk going to transmit this fax using t38 ? PSTN FAX MACHINEASTERISK(1.6.0b2) FXO CARD---t38?ATA/Gateway-FAX MACHINE 2 - If the first answer is yes, if we compile app_fax with asterisk 1.4x same behavior could be achieved ? Regards, Fernando Tzafrir Cohen wrote: On Mon, Feb 25, 2008 at 05:32:24PM -0300, Fernando Berretta wrote: Dear All, Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. and will be able to receive faxes and negotiate with voip CPE's like ATA's to transmit faxes which comes from FXO cards to VoIP Devices using T38 ? it is possible to compile this version of app_fax to work with Asterisk 1.4x ? Someone has tried it ? You have rx_fax for 1.4 . You also have fax detection in chan_zap, and thus can send faxes from the PSTN to rx_fax. Not exactly the same, but maybe this is actually what you're looking for. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Thanks for clarify.. so Asterisk will be able to receive faxes which comes from a Gateway using t38 but will not be able to relay faxes which comes from PSTN through a FXO card to other Gateway using t38 can this version of app_fax be used with Asterisk 1.4x ? Steve Underwood wrote: zoa wrote: T.38 will not work with the fxo card. Zoa That statement is a bit vague. What has been put in add-ons so far is only support for T.38 termination. Not T.38 gateway operation. Steve Fernando Berretta wrote: Dear All, Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. and will be able to receive faxes and negotiate with voip CPE's like ATA's to transmit faxes which comes from FXO cards to VoIP Devices using T38 ? it is possible to compile this version of app_fax to work with Asterisk 1.4x ? Someone has tried it ? Best Regards, Fernando Thomas Kenyon wrote: Steve Underwood wrote: I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 version of app_fax (and a few small changes in 1.6.0b4), which I thought someone would have mentioned to you, since it does use spandsp. (Or at least the configure script checks for spandsp, I haven't actually looked at the code). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Dear All, Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. and will be able to receive faxes and negotiate with voip CPE's like ATA's to transmit faxes which comes from FXO cards to VoIP Devices using T38 ? it is possible to compile this version of app_fax to work with Asterisk 1.4x ? Someone has tried it ? Best Regards, Fernando Thomas Kenyon wrote: Steve Underwood wrote: I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 version of app_fax (and a few small changes in 1.6.0b4), which I thought someone would have mentioned to you, since it does use spandsp. (Or at least the configure script checks for spandsp, I haven't actually looked at the code). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO Cards - T38
Hi, Could some one let me know if a fax is received through a FXO card connected to * and fax machine is connected to a Linksys FXS device which support T38, is T38 going to be used for faxes which comes from PSTN or go through PSTN ? or because of Asterisk T38 passthrough support it is not possible ? so is for this scenery better to use external FXO gateways with t38 support ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External Incomming Call Directed PickUP
Someone could please help me ?
Regards,
Fernando
Fernando Berretta wrote:
Dear Lacy,
We are using Standard FreePbx installation and we are trying to direct
pickup all the calls with **EXT NUMBER.
[app-pickup]
include = app-pickup-custom
exten = _**.,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten = _**.,n,Pickup(${EXTEN:2})
This is the FreePbx configuration for call pickup but doesn't work for
calls which comes from users which are in other context neither
incoming calls from-trunk
Any help will be appreciated
Regards,
Fernando
Lacy Moore wrote:
My magic orb is on the fritz. Can you give some more info? What
extension is ringing? What are you dialing to pick up? What does
your conf files look like?
I think I might know what the problem is, but I need a little more
info. Read core show application Pickup carefully, and then re-read
it 3 or 4 more times. It seems odd at first, but then you catch on.
You are picking up the calling channel, not the called extension.
On Jan 25, 2008 5:28 PM, Fernando Berretta [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with Directed PickUn and Asterisk 1.4.
Directed call pickup **EXT works ok with internal calls which are in the
same CONTEXT but,, with calls in which are from other context or
incoming calls from IVR this function doesn't work as is pointed in
http://bugs.digium.com/view.php?id=11639
I'm using FreePbx 2.3,, and dont know how to solve or workaround this
problem
Could some one please help me.
Best Regards,
Fernando
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Re: [asterisk-users] External Incomming Call Directed PickUP
Dear Lacy,
We are using Standard FreePbx installation and we are trying to direct
pickup all the calls with **EXT NUMBER.
[app-pickup]
include = app-pickup-custom
exten = _**.,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten = _**.,n,Pickup(${EXTEN:2})
This is the FreePbx configuration for call pickup but doesn't work for
calls which comes from users which are in other context neither incoming
calls from-trunk
Any help will be appreciated
Regards,
Fernando
Lacy Moore wrote:
My magic orb is on the fritz. Can you give some more info? What
extension is ringing? What are you dialing to pick up? What does
your conf files look like?
I think I might know what the problem is, but I need a little more
info. Read core show application Pickup carefully, and then re-read
it 3 or 4 more times. It seems odd at first, but then you catch on.
You are picking up the calling channel, not the called extension.
On Jan 25, 2008 5:28 PM, Fernando Berretta [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with Directed PickUn and Asterisk 1.4.
Directed call pickup **EXT works ok with internal calls which are in the
same CONTEXT but,, with calls in which are from other context or
incoming calls from IVR this function doesn't work as is pointed in
http://bugs.digium.com/view.php?id=11639
I'm using FreePbx 2.3,, and dont know how to solve or workaround this
problem
Could some one please help me.
Best Regards,
Fernando
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[asterisk-users] External Incomming Call Directed PickUP
Hi, I'm having problems with Directed PickUn and Asterisk 1.4. Directed call pickup **EXT works ok with internal calls which are in the same CONTEXT but,, with calls in which are from other context or incoming calls from IVR this function doesn't work as is pointed in http://bugs.digium.com/view.php?id=11639 I'm using FreePbx 2.3,, and dont know how to solve or workaround this problem Could some one please help me. Best Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Julio, Thanks for your suggestion, at this stage I would like this version of g729 running in my box.. but,,, is good to know the paid version works without any problems in this machine for the next stage. Best Regards, Fernando Julio Arruda wrote: Fernando Berretta wrote: Dear Mindaugas, Thanks for your promt response I've already tried this but.. it's not working,, what file do you think I should use ? any other idea ? Fernando, I've used the official/legal G729 codec sold at www.digium.com in Athlon boxes w/ asterisk 1.4 without problems, have you tried this option ? Mindaugas Kezys wrote: Rename to codec_g729.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so Copy to /usr/lib/asterisk/modules chmod 777 codec_g729.so restart Asterisk show translations Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Fernando Berretta *Sent:* Monday, November 26, 2007 6:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working Thanks, Fernando Mindaugas Kezys wrote: For testing purposes you can try one of these: http://kvin.lv/pub/Linux/Asterisk/ Mindaugas Kezys http://www.kolmisoft.com Advance Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Fernando Berretta Sent: Friday, November 23, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Hi, I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor 4000+ but.. all packages I've download haven't worked. Could someone please let me know what package should I download ? Best Regards, Fernando [EMAIL PROTECTED] modules]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4222.52 processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4219.18 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Dear Mindaugas, Thanks for your promt response I've already tried this but.. it's not working,, what file do you think I should use ? any other idea ? Best Regards, Fernando Mindaugas Kezys wrote: Rename to codec_g729.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so Copy to /usr/lib/asterisk/modules chmod 777 codec_g729.so restart Asterisk show translations Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Fernando Berretta *Sent:* Monday, November 26, 2007 6:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Hi, I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor 4000+ but.. all packages I've download haven't worked. Could someone please let me know what package should I download ? Best Regards, Fernando [EMAIL PROTECTED] modules]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4222.52 processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4219.18 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Live
Hi, Philipp von Klitzing posted this solution in Dec. 2005 Answering machine mimic: Listen while caller is leaving voicemail for you; with pick-up option Is there any other way to listen while caller is leaving a voicemail for you? Thanks Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is overwriting proxy Via Header
Hi, I having a problem with my asterisk, it is overwriting the Proxy Via header with its own ip address and answering to the Proxy with the modified header, so the Proxy is having problems to route the response. I've tried with different versions of asterisk and nothing is changing, and if I try in other Server all works perfect, the problem is related with this particular server running over Linux dit_rs_poa_mtz_gw1.local 2.6.18 #1 SMP PREEMPT Fri Sep 22 10:43:25 BRT 2006 i686 i686 i386 GNU/Linux The scenary is like this: IPPhone---Proxy1--Asterisk Invite sent by the IPPhone INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b From: teste sip:[EMAIL PROTECTED];tag=d772c33c63ebf84c To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:22149 Supported: replaces Proxy-Authorization: Digest username=5551125, realm=200.X.X.136, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0001, cnonce=eed75407c0d78607, opaque=4c4f15e2744c43bb0790c60a78c00552, nonce=453778dd3e3c605897e1efdeb823fc53122bc50c, response=67cab99628290773609250e828628f14 Call-ID: [EMAIL PROTECTED] CSeq: 63837 INVITE User-Agent: Grandstream BT110 1.0.8.23 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 388 Invite sent by the Proxy to * INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Contact: sip:[EMAIL PROTECTED]:50386 CSeq: 63837 INVITE From: teste sip:[EMAIL PROTECTED]:5060;tag=d772c33c63ebf84c Proxy-Authorization: digest username=5551125, realm=200.X.X.136, nonce=453778dd3e3c605897e1efdeb823fc53122bc50c, cnonce=eed75407c0d78607, response=67cab99628290773609250e828628f14, uri=sip:[EMAIL PROTECTED], opaque=4c4f15e2744c43bb0790c60a78c00552, qop=auth, nc=0001, algorithm=MD5 To: sip:[EMAIL PROTECTED]:5060 Via: SIP/2.0/UDP 200.X.X.136:5060;branch=z9hG4bKbced0281e38aa078 Via: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b;received=201.X.X.212; rport=50386 Record-Route: sip:200.X.X.136:5060 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: Grandstream BT110 1.0.8.23 Call-Id: [EMAIL PROTECTED] Max-Forwards: 70 Content-Length: 389 supported: replaces content-type: application/sdp Trying sent by Asterisk with via modified SIP/2.0 100 Trying v: SIP/2.0/UDP 200.X.X.131:5060;branch=z9hG4bKbced0281e38aa078;received=200.X.X.136 v: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b;received=201.X.X.212; rport=50386 f: teste sip:[EMAIL PROTECTED]:5060;tag=d772c33c63ebf84c t: sip:[EMAIL PROTECTED]:5060 i: [EMAIL PROTECTED] CSeq: 63837 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY m: sip:[EMAIL PROTECTED] l: 0 Could someone please tell me why asterisk is replacing proxy ip address with its own ip address in the last one via header ?? How can I solve it ? Regards, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CTI
Hello, Have someone implemented * like as a CTI platform with IVR, VoiceMail, Fax to tiff files, etc etc using Digium/Sangoma Dual T1/E1 interface cards? Does it work ok? Is the audio quality good when all the ports are in use? Is there any issue regarding on faxes on digital trunks? How do you suggest implementing this kind of solution? Best Regards, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE : [Asterisk-Users] TDM 2400 With 24 FXO
Dear Francois, Thanks for your advise,, I'll buy the echocan module Best Regards, Fernando -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 18, 2006 6:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] TDM 2400 With 24 FXO Hello Fernando, I have checked this card with and without hardware echocan : the hardware echocan module does the job better than the zaptel software can do it. I recommand this module without any doubt. But, the echocan algorithms in zaptel are better and better and the CPUs power grows permanently. It is possible to use this card without hardware echocan, but you will encounter the same results, in this case, as you can obtain with the other TDM Digium's cards : correct for certain situations, not for all extreme cases, depending what listening level your users want, lines specifications and what critical echo threshold they can admit before to not be able to do correctly their job. Near same thing for E1/T1 harware echocan features. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Fernando BERRETTA Envoyé : vendredi 17 mars 2006 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM 2400 With 24 FXO Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM 2400 With 24 FXO
Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 DTMF Caller ID
Hello, I'm not detecting caller Id through DTMF with my TDM400 card and don't know how to put this to work. Could someone please help me ? How should I configure * to achieve this? Thanks, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
