similar to: EC needed in all-digital situation?

Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hello, My company has invested heavily in Counterpath?s Stretto provisioning platform for Mobile and Desktop VoIP clients . At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , which many of our users require. Their provisioning system assumes that both devices will use the same SIP extension for auth however. Normally

Version: dovecot --version: 1.0.0 OS: OS X - 10.4.9 Architecture: G5 Dual/1.8Ghz Had a fairly normal time correction last night: May 9 20:56:56 G518X2 ntpd[219]: time set -1.212733 s Which dovecot duly noted: May 9 20:56:56 G518X2 dovecot: auth(default): Time just moved backwards by 1 seconds. I'll sleep now until we're back in present. This morning all of the dovecot processes are

I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but

In the process of turning up a new pri. Zttool indicates the T1 is ready with no alarms. asterisk*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange

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What's available to remotely monitor services? What I'd like is something that can run scripts for each service to connect to a port and verify that it's up, and then send me an SMS message (phone text) to let me know which, if any, are down. Also, does a script exist that checks all the services listed by chkconfig and reports those that should be up but are down?

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

Hello, My company has been asked to help with the upgrade of several Freebsd systems that are pretty old. The customer is running a file server samba also running apache running FBSD 4.2, he wants to upgrade using cvsup & the make buildworld procedure to upgrade to 5.x. Im very familier with the make buildworld procedure however there have been significant changes between 4.2 & 5.x so is

Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. That being said, they are also alot more money than the Rhino cards and my friend currently has 1 digium 4 fxo card in their system and they need to add another phone line, plus they have echo problems and quality

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

A friend of mine made the following comment in a discussion I had with him that on a website we adminster we should offer a) WAV or maybe shorten files b) Ogg as a decent reference lossy encoded version He's been trying to convince me that we should offer MP3 (in lieu of WAV) and possibly Ogg. The audio files are primarily vocals I am not a physics guy but his statements don't