search for: yourownisp

...k config is the same regarding NAT for this extension and I have the Sipura registering with * so I am at a loss as to why Asterisk loses or stops ringing this device. I have dug around and can't seem to solve this issue so far, any help would be appreciated. -- Start Your Own ISP! http://www.YourOwnISP.com

...ps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com

...there a way to make the above config work and go to the * voicemail after 20 seconds if the called party does not answer after 20 seconds? Also, what happens if the called party's line is busy, have not run into this yet so I am curious. Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041228/f0413a53/attachment.htm

...9;s designed for a SOHO environment, we are doing a bit more than that. Will this thing hold up? Can it be trusted? Anyone using this for QOS and Asterisk in a production setup. Any thoughts or suggestions or warnings would be appreciated! Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

.... Basically I want one Asterisk server to be the traffic cop and send some calls directly to ATA's and some calls to another Asterisk server, the other Asterisk server will then direct the calls to the end users ATA on their desk or to vmail etc. Thanks.. -- Start Your Own ISP! http://www.YourOwnISP.com

...nd the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has certain redundant features with an ISDN line which are needed. Any help here would be appreciated! -- Start Your Own ISP! http://www.YourOwnISP.com

...usable services. Can someone recommend a termination partner for our VOIP Venture that can provide reliable services, good features/DID's and GOOD customer service? Price is important as well but comes last in line after the items mentioned above. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

...the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com

Hi, I am using a Netjet-s ISDN Card, and am having some trouble dialling out (Incoming Works Fine). TRUNK=Modem/ttyI0 exten => _90XXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _90XXXXXXXXX,2,Congestion I get the following when diallingout: -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Modem/ttyI0/04XXXXXXXX||Ttm") in new

...onfig is the same regarding NAT for this extension and I have the Sipura registering with * so I am at a loss as to why Asterisk loses or stops ringing this device. I have dug around and can't seem to solve this issue so far, any help would be appreciated. -- Start Your Own ISP! http://www.YourOwnISP.com

...te is less complete and polished. VoicePulse DID's are active immediately I point anybody who asks to Sixtel. On Mon, 13 Dec 2004, Me wrote: > Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. > > Thanks! > > -- > Start Your Own ISP! > http://www.YourOwnISP.com > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >

...CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. Any help would be appreciated. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

...via IAX. Also, what does it mean when I see "hangup" in the CRD entry? Does it mean that our caller (our extension) hungup or that the called party hung up or?? Sound like stupid questions, I know but none the less I would like to know the answers :) -- Start Your Own ISP! http://www.YourOwnISP.com

...xy" field so no incoming calls get to the SPA anymore. It seems that the Outbound Proxy field overrides the Proxy field, are they one in the same? If so, what's the purpose? Any help on getting this to work would be greatly appreciated. Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com

Weird Problem: I have 2 EXT. One will ring and NOT go into VM (eventually call will timeout/hang up), the other EXT goes into VM when the call is not answered like it should. If I enable DND, then the call will go directly in VM as it should. Any ideas what it might be? Thank you, Jake

...NY options technically when ordering this PRI (in the US) but since this is the first time ordering a voice circuit I am clueless as to what options we need. Any clues would be helpful or maybe something has already been written about this? Thanks in advance! -- Start Your Own ISP! http://www.YourOwnISP.com

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

...W Drive w/Burning Software f.. 3.5" 1.44MB Floppy Drive g.. ATI Rage XL with 8MB Onboard h.. Onboard RAID controller i.. (2) Intel Ethernet Controllers (1x1000BT Gigabit & 1x10/100) j.. 2U Rackmount Chassis w/ 500-Watt Power Supply Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041228/107f0804/attachment.htm

Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the