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hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of

...ost should be about the same overall(depending on which channelbanks you buy), except you will be able to use a much smaller server case with the quad T1 card. The most we have done is two channelbanks off of a quad T1 card in a single machine and it works just fine. MATT--- On 2/15/06, maka <icokan@gmail.com> wrote: > hello, > > I am planning a fairly large hotel VoIP system, using analog phones. It will > consist of about 100 analog phones, that must have access to a VoIP server. > I am considering an option to use a couple of asterisk boxes, bundled with a > total of fo...

Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller

Hi all, I have been looking at some billing solutions for asterisk. I saw there is Trabas VoIP Billing which apparently is working through radius cdr records, and also astPP which was recently released, and CDRTool. Has anyone been able to succeffully use radius with asterisk for CDR records? I tried app_radius with freerdius according to the wiki docs, but the Radius(CPP) keeps playing a

Hi all, Can I choose a different source ip address that asterisk uses when sending registration requests, and another one when transfering calls to a destination with the DIAL() command? The thing is, when I register with my sip provider proxy at let's say 111.111.111.111, I want the registration packets to be sourced from one of my ip addresses - 222.222.222.222 for example. I will then use

Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to