search for: eaperezh

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition?

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) Hi, im running asterisk as another user (user x group x), While trying to attach to the co...

...<billings at negate.org> > _______________________________________________ > CentOS mailing list > CentOS at centos.org > https://lists.centos.org/mailman/listinfo/centos -- --------------------- Erick Perez Quadrian Enterprises S.A. - Panama, Republica de Panama Skype chat: eaperezh WhatsApp IM: +507-6675-5083 ---------------------

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

When is supposed to be released the Zimbra+Asterisk version? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 mailto: eaperezh@gmail.com mailto: eperez@ptysystems.com ------------------------------------------------------------

I cant find the server cd iso or torrent. Was it removed from all the mirrors? thanks, -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

Hi, I am about to give a training session in Linux, using Centos 4.4 as a base. I want to give my students (MS windows administrators) printed books about Linux and hopefully Centos based. It is a total of 20 books. I'm also interested in purchasing 20 copies of the DVD of Centos 4.4 x86/x86_64 distro. My initial search at amazon returned no useful results. It can be RHEL 4, Centos 4 or

...heeva.com/ for a commercial Asterisk call center solution that offers a ton of functionality for a price. That's about all I know of for full Asterisk call center suites. Let me know if you have any questions about astGUIclient. MATT--- -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Tuesday, March 15, 2005 10:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Center software opensource or commercial Hi there, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, ma...

Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with

Hi All I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone. However I cannot get the incoming calls to hit the asterisk box. Any help will be appreciated. Thanks, Lal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060612/36de60cc/attachment.htm

Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big

...> Mob. : 06 51 80 12 12 > _______________________________________________ > CentOS mailing list > CentOS at centos.org > https://lists.centos.org/mailman/listinfo/centos > -- --------------------- Erick Perez Quadrian Enterprises S.A. - Panama, Republica de Panama Skype chat: eaperezh WhatsApp IM: +507-6675-5083 ---------------------

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary.

...ter/doc/manual-install.md)? > _______________________________________________ > CentOS mailing list > CentOS at centos.org > https://lists.centos.org/mailman/listinfo/centos > -- --------------------- Erick Perez Quadrian Enterprises S.A. - Panama, Republica de Panama Skype chat: eaperezh WhatsApp IM: +507-6675-5083 ---------------------

...unroll-loops # below is another 5% faster or so. CFLAGS += -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant # this is fun for PPC #CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic # this is fun for x86 # The line below was commented by Erick Perez eaperezh@gmail.com #CFLAGS += -march=pentium3 -msse -mfpmath=sse,387 # adding -msse -mfpmath=sse has little effect. #CFLAGS += -O3 -msse -mfpmath=sse #CFLAGS += $(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi) CFLAGS += $(sh...

...destroy all copies of it." > _______________________________________________ > CentOS mailing list > CentOS at centos.org > https://lists.centos.org/mailman/listinfo/centos -- --------------------- Erick Perez Quadrian Enterprises S.A. - Panama, Republica de Panama Skype chat: eaperezh WhatsApp IM: +507-6675-5083 ---------------------

Do anybody could compile app_conference on x86_64??? I tryied with two versions of app_conference and got the same problem on compiling: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared recompile with -fPIC app_conference.o: could not read symbols: Bad value" ENVIRONMENT:

...on in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detec...

...ntOS mailing list >> CentOS at centos.org >> https://lists.centos.org/mailman/listinfo/centos >> > > > -- > > --------------------- > Erick Perez > > -- --------------------- Erick Perez Quadrian Enterprises S.A. - Panama, Republica de Panama Skype chat: eaperezh WhatsApp IM: +507-6675-5083 ---------------------