search for: milioto

...- Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do you suggest according I said above? Is there another device which deserves attention? Thanks very much in advance, Sebastian ---------------------------------------------------- Sebastian Milioto ITC Cid Campeadro 440 Rio Tercero, Cordoba, Argentina msn: sebamilioto at hotmail.com ---------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100210/62990bbe/a...

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian

Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

...had happen to other people, however I can't find how they solved it. It seems to be a codec problem.. however I've already tried configuring g729a,g711u, and g711a in spa3102 with no success.. Can anybody help me with that, please? Sebastian On Fri, Mar 19, 2010 at 10:16 AM, Sebastian Milioto <smilioto at gmail.com>wrote: > Thanks! > > > On Thu, Mar 18, 2010 at 5:04 PM, Joseph <syscon780 at gmail.com> wrote: > >> On 03/18/10 16:22, Sebastian Milioto wrote: >> >Somebody has 5.1.7 firmware for SPA3102? >> >I'm having issues with inbo...

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

...ut the only I get is Allison telling me "circuit busy now, please call again later" or some thing similar. I'm trying make it by AMP and editing sip.conf and extension.conf, and I read all about it in voip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:smilioto@GMAIL.com IM: sebamilioto@hotmail.com

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Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

...iste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 13 > Date: Fri, 21 Nov 2008 11:59:29 -0300 > From: "Sebastian Milioto" <smilioto at gmail.com> > Subject: [asterisk-users] Ping > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <e6e7910f0811210659m7dc9d8b7t4c171a9093b59c95 at mail.gmail.com...

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hi all, do you know any firmware release which fixes that issue for cisco ATA186? ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent with a completely bogus URI, making it impossible to cancel a call. bug no workarounds This is from: http://www.communigate.com/SIP/SIPProblems.html. In other words, you make a call, other side doesn't pickup, then you hang up,