similar to: Sip provider group

I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination provider. Im tring to start a small Pre-paid long distance service. Thanks Erick

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP

Hi, Im trying to register to a SIP provider that told me that they only need to authenticate using IP. the following string generates response 405 register => asteriskIPaddress@SIPproviderIP:5060 doing the following is not alowed by asterisk register => @SIPproviderIP:5060 any ideas? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

Hello, I get warnings in my asterisk log: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call. I've used sip debugging to figure out the cause. It's my D-link DVG-1120S that don't understand message-summary events that asterisk sends out for MWI indication to the client. Is there any way to disable this in asterisk for this particular client? Tanks in advance, Magnus

Here is a runable program. When I plot Day and Wgt, it graphs all the data points. All I need is daily.sub1 plotted. I also need each "Tanks" to have its own col or pch. When I run it with the line with pch, it gives me nothing. rm(list=ls()) Trial<-rep(c(1,2),each=12) Tanks=rep(c("a3","a4","c4","h4"),each=3,2) Day=rep(c(1:12),2)

OK, I need help plotting. I have column headings of Day, Wgt, Foodin, Rep, Grp and Tanks. Rep=c(1,2,3) and Tanks=c(a1,a2,a3,a4,a5,a6, c1,c2,c3,c4,c5,c6, h1,h2,h3,h4,h5,h6). I created a subset where I only would like Rep=2, and Tanks=c(a4,c4,h4) and would like to graph (points) of Wgt and Day. I would think that I only need 3 colors, but when I run with only 3, only 2 lines show up. When I add

Hi, I have a customer with a sun cobalt running Sendmail 8.10.2/8.10.2 and we are phasing out the sun cube due to some limitations. So we have installed a new centos 5.x server. the format of our current emails are username at domain.com and the new format will be name.lastname at domain.com. We have 1600 accounts. Both server are in the LAN. The MX record that the world sees, point to our

Hello R experts, I have having a difficult time figuring out how to perform and interpret an ANCOVA of my nested experimental data and would love any suggestions that you might have. Here is the deal: 1) I have twelve tanks of fish (1-12), each with a bunch of fish in them 2) I have three treatments (1-3); 4 tanks per treatment. (each tank only has one treatment applied to it) 3) I sampled

The following strange conditions is happening while I try to dial a SIP user from another SIp user. SIP to Zap dialing is fine, as all 4 users can call PSTN. I'm using Asterisk SVN-branch-1.2-r51359M Example: extension 3210 calls extension 3213. They are all registered properly: chrom01*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 3213/3213

hello everyone, I've looked through Google and this list's archives but have come up empty. here's may question: I have an NT laptop where I log in as administrator, I have a linux box with an account of my one (erick). Everytime I try to access the filesystem on the Linux box, NT sends credentials for administrator which of course, is wrong. when the challenge fail I'm prompted

Dear all, I have tried to calculate a GLMM fit with lmer (lme4) and glmmPQL (MASS), I also used glm for comparison. I am getting very different results from different functions, and I suspect that the problem is with our dataset rather than the functions, but I would appreciate help in deciding whether my suspicions are right. If indeed we are attempting the wrong type of analysis, some

I am transitioning from SAS to R and am struggling with a relatively simple analysis. Have tried Venables and Ripley and other guides but can't find a solution. I have an experiment with 12 tanks. Each tank holds 10 fish. The 12 tanks have randomly assigned one of 4 food treatments - S(tarve), L(ow), M(edium) and H(igh). There are 3 reps of each treatment. I collect data on size of each

Hi Hope someone can help :) I am testing 4 PSTN termination providers. 3 SIP and 1 IAX IAX and 1 of the SIP providers work fine. Now the wierdness: 2 SIP providers I can only get oubound calls to ring at the destination and then nothing more. 1 gets as far as SIP code 183 (and ringing on the src handset ...yay) the other doesn't get past 100. Added to this inbound calls

When we make calls out of asterisk to the PSTN via a SIP termination service provider the called party gets a slight echo of their voice. Here is the setup; analog phone <> Linksys ata <> asterisk <> sip provider sonus GSX 9000 <> PSTN <> called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight

I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: