similar to: ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min

starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us support at cdsportal.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD

We would like to use Asterisk to deploy VoIP to our broadband internet access customers. Which VoIP providers (that are reliable & stable) provide wholesale DID's?

Looking for a good termination provider for US/Canada Please contact offlist. Shady -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061219/75bb266e/attachment.htm

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Hi List, I have spent more than 30 minutes, but failed to read in this file using the read.table() function. I could not figure out how to fix the following error. Error in scan(file, what, nmax, sep, dec, quote, skip, nlines, na.strings, : line 1 did not have 6 elements Any help would be be appreciated. Thanks, Pradip Muhuri ####### below is the reproducible example xd1 <-

Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and

Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they "think" packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hello List, I'm trying to help a family from the Dominican Republic and to do so need a temporary DID from DR. The short story is that there is a 2 year old here with a serious heart defect from a remote area of Dominican Republic near the Haitian border. He was referred to Gift of Life D.R. who eventually contacted Gift of Life Central Florida and this is where I'm involved. The child

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Dear All, I am trying to recieve call from inbound proxy then route to internal peer (localhost) and then route to outgoing sip proxy but it failing with subject error. Can any one please correct me what i am doing wrong in below config. SIP.conf [Inbound] type=peer context=introuting host=184.107.XXX.XXX disallow=all allow=all [astinside] type=peer context=introutingB host=localhost

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. --------------

Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hi list, My Asterisk config for outgoing calls is the following: exten => s,1,Dial(SIP/${MACRO_EXTEN}@voipprovider,60,g) exten => s,n,GotoIf($[\"${ANSWEREDTIME}\" = \"\"]?pri:hang) exten => s,n(pri),NoOp(Problems with voip provider trying PRI) exten => s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g) exten => s,n(hang),HangUp in most cases it works well but, if my