RE: [Asterisk-Users] Asterisk hobby box
> From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Russ Price > Sent: Wednesday, November 16, 2005 2:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk hobby box > > Funny you would say that - I have a box running with a pair of X100P > clones at the moment. I did have to tinker with transmit and receive > levels, but, since then, they have run just fine on my old 533 MHz > Celeron box, with no echo problems on 1.2.0 beta2 or rc2. I > have them > hooked into the voice mail ports on a Panasonic KX-TA624, and > managed to > integrate it in as a voice mail system. > > The main advantage for the TDM400 is that you can add FXS > ports, or you > could have four FXOs if you needed them. Still, the TDM400's cost is > rather steep for a "hobby" box. > > I wouldn't recommend trying to use more than one X100P card > unless you > can insure they get separate IRQs, and going with more than two would > probably not be a good idea. > > Russ Hi, I am not surprised you managed to sort out your echo problems quickly and easily. You see, the Panasonic KX-TA624 behaves the same every time, while a PSTN line would have a varying quality, noise, echo every time, depending on the other end sometimes. Yiannis. ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] Asterisk hobby box
Hi, the best thing to do is get a Sipura 3000 that has 1 FXO and 1 FXS port. You won't need to bother with IRQs and echo problems that at least here in UK we have with FXO cards. Yiannis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Logan Sent: Wednesday, November 16, 2005 1:02 PM To: Asterisk-Users List Subject: Re: [Asterisk-Users] Asterisk hobby box Hi everyone! Okay. I was reading on the voip-info.org about FXO and FXS. Is it possible just to get a card with FXO and FXS together? I know Digium sells them, but as I've said, I'm looking to spend too much. Thanks for everyone's input! Logan. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] Re: www.openpbx.org
Personally, I believe it's a good thing. It gives more choice. Look at other products: IPCop (Linux based firewall) is a fork derived from Smoothwall. They made such a nice job that Smoothwall were playing catch-up with IPCop for quite some time. I don't know the current situation. GPL allows forking and forking is a form of evolution. YC > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Jon Pounder > Sent: 07 October 2005 17:20 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: www.openpbx.org > > > > > On Friday 07 October 2005 11:28, Jon Pounder wrote: > >> contributors more choice. As long as the two streams stay compatible > >> (which they likely will) it should be better for everyone. > > > > Don't count on it, the rumblings in the IRC channel sound like > it will be > > totally INcompatible except to pass calls between. > > and if there is a big enough community that wants to stay compatible, > there is nothing to stop even more forks. > > Who would have thought there would be support for so many linux > distributions ? phones are just as common as linux servers so wouldn't you > think there would be the resources out there to support at least several > flavours of an open source pbx ? > > > > > -A. > > ___ > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > Jon Pounder > >_/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ > _/_/_/ _/ _/ _/_/_/ _/ _/_/ >_/_/ _/_/ _/ _/_/ _/_/ _/ > _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ > > > Inline Internet Systems Inc. > Thorold, Ontario, Canada > > Tools to Power Your e-Business Solutions > www.inline.net > www.ihtml.com > www.ihtmlmerchant.com > www.opayc.com > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] a simple call to my girlfriend
No it wouldn't. It doesn't run on his laptop. I would suggest two FWD accounts with two SIP softphones. Unless you really want to go the Asterisk way and use IAX softphones over NAT that works slightly better. Yiannis. > -Original Message- > > Skype would do you the best. > > On 6/2/05, Hendrik Wouters <[EMAIL PROTECTED]> wrote: > > Hi, > > > > Some background: > > > > I would like to call my girlfriend over the internet. We are > both behind a nat > > router and I want to avoid portmapping. > > I've heard that you can call someone behind a firewall (nat > router) with the > > IAX protocol, but I'm not sure. > > > > The questions: > > > > Do I have to set up my own PBX asterisk server or are there any > other (free) > > servers where I can register on and connect to? > > > > Which is the best (linux) client to call someone with IAX? > > > > Thanks in advance > > > > Greetings Hendrik > > > > P.S. I don't want to use skype (not open standard, it still > doens't work well > > in Linux and eats al the time of my old laptop CPU). > > ___ 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
[Asterisk-Users] Satellite Providers
Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ 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
RE: [Asterisk-Users] Router Recommendations Please
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Michael > Graves > Sent: 19 January 2005 14:06 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Router Recommendations Please > > > On Tue, 18 Jan 2005 20:01:51 -0600, [EMAIL PROTECTED] wrote: > > >Hello all, > > > >We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly > >CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will > >increase to 4/4 next year. > > > >The Netopia simply breaks out our WAN IPs, and we've got a > switch hooked up > >to it on the inside (Actually I've got a QoS box in-between). > > > >- > >| Internet | > >| on Cat5 | > >- > > | > >- > >| Netopia | > >- > > | > >- > >| QoS Bridge | > >-- > > | > >-- > >| switch for WAN IPs | > >-- > > | | | > >-- > >| LAN Switch | > >-- > > | | | | > > > > > >Any recommendations on something that isn't as pricey as Cisco? I'm in > >discussions about us building Linux units down the line, but for now we > >need something we can buy. Cisco is too expensive for us. > > > > m0n0wall is my favorite. > > web site http://m0n0.ch/wall/ > > Review on Tom's Networking > http://www.tomsnetworking.com/Reviews-161-ProdID-MONOWALL.php > > Open source running on either a plain vanilla PC or a Soekris embedded > platform. > > Michael > I vote m0n0wall too. It does what you expect it to do and more! Yiannis. ___ 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
RE: [Asterisk-Users] X100P red alert
Yes, it's true! Connect the card to a phone line and the Red Alert disappears. I don't think it draws power from the phone line, but it gives a red alert if the phone line is not there. I have experienced it myself! Yiannis. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Alex van Es > Sent: 19 October 2004 20:56 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] X100P red alert > > > Hi all, > I just got my X100P card installed and asterisk keeps on complaining > that it cannot create a zap channel. I read somewhere on the internet > that the zap card will not work when a phoneline is not plugged in, > cause is draws power from the phoneline. Is this correct? Of course > eventually > I will connect it to a phoneline, but I would just like to know this > for sure.. > > Alex > > -- > Alex van Es - [EMAIL PROTECTED] > http://photography.icepick.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over IP doesn't works
Well, assuming that some of these CODECS do error correction and drop any information that hasn't come through instead of doing error detection and request to re-transmit the lost information, is somewhat expected. Are there any Fax over IP protocols? Yiannis. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Pedro Howat > Rodrigues > Sent: 19 October 2004 15:53 > To: Miroslav Nachev; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] Fax over IP doesn't works > > > Hi , > > I tried this a lot, but with no sucess , even in a local network , there > is always some loss and you receive only chunks of the original file . > > Pedro. > > Miroslav Nachev wrote: > > > Hi, > > > > We try to send Fax through IP Network but without success. The > >other party use NetCentrex SoftSwitch and our communication protocol > >between us is H.323 (OpenH323). The error that the other party receive > >is: "bearer capability not imoplemented". > > > > Is it possible to send Fax using Asterisk to the other party > >through IP network? What T.38 and Asterisk? > > > > > > Regards, > > Miro. > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for recommendations for a low-cost FXO toIP gateway.
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Mark > Sent: 15 October 2004 02:41 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Looking for recommendations for a low-cost FXO > toIP gateway. > > > At first I thought the X100P was what I was looking for, but now it > looks to me like the X100P does not have an IP interface, so it would > require all audio to run through the CPU. I'm familiar with ATA186's, > which I think are comparable to the IAXy box, and I'd just like to find > something like that that provides an FXO interface. Can anyone help me? As far as I know there are quite few companies out there that do FXO Gateways. Mediatrix does a 2-port and 4-port anologue FXO model. The other solutions is a Sipura-3000. The Sipura 3000 will need it's own dialplan too, because it has an FXO, FXS and Eth port. So, you will have to do some configuration. Some people have reported different issues with it. one of them is over-heating. I haven't used any of the Mediatrix products, or any other FXO gateway, so I don't have personal experience. I hope it helps, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ringing for SipToneII
Hi, I have a couple of IpDialog SipToneII phones and although I understood that they had a choice of 5 ringtones, it turns out that it is Distinctive Ringing. I contacted IpDialog support and sent me this. ---snip-- The phones you have support 5 different ringtones and 4 call waiting tones. You do not need new firmware. Please see attached document. The phone will ring differently depending on the contents of the Alert-Info header sent to the phone in the SIP INVITE. Some have used this feature on our phone to create two virtual lines on the phone. If you create two aliases (phone numbers) for the same phone in your PBX, and send a different Alert-Info header for each phone number, the phone will ring differently for each "line." Here's an example of how to use Alert-Info: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.68.202.101:5065 From: "SIPTool" ;tag=ABCD To: "Nobody Nowhere" Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Alert-Info: Contact: "SIPTool" Content-Length: 130 Content-type: application/sdp ---/snip-- They also sent me this (I don't know if it of any help ---snip- Bellcore priority and Call Waiting tones per GR-506-CORE and GR-526- CORE Bellcore priority 1 - standard phone ring cycle Call Waiting Tone used with this Ring: Call Waiting priority 1 Alert Info string used : Number of States = 3; State1 = ToneOn; State1 Length = 2000; State1 Freq1 = 725; State1 Freq2 = 750; State2 = ToneOff; State2 Length = 4000; State3 = ToneRepeat; State3 Reps = 0; Bellcore priority 2 Long-Long Call Waiting Tone used with this Ring: Call Waiting priority 2 Alert Info string used : Number of States = 5; State1 = ToneOn; State1 Length = 800; State1 Freq = 725 State1 Freq2 = 750; State2 = ToneOff; State2 Length = 400; State3 = ToneOn; State3 Length = 800; State3 Freq1 = 725; State3 Freq2 = 750; State4 = ToneOff; State4 Length = 4000; State5 = ToneRepeat; State5 Reps = 0; Bellcore priority 3 Short-Short-Long Call Waiting Tone used with this Ring: Call Waiting priority 3 Alert Info string used : Number of States = 7; State1 = ToneOn; State1 Length = 400; State1 Freq1 = 725; State1 Freq2 = 750; State2 = ToneOff; State2 Length = 200; State3 = ToneOn; State3 Length = 400; State3 Freq1 = 725; State3 Freq2 = 750; State4 = ToneOff; State4 Length = 200; State5 = ToneOn; State5 Length = 800; State5 Freq1 = 725; State5 Freq2 = 750; State6 = ToneOff; State6 Length = 4000; State7 = ToneRepeat; State7 Reps = 0; Bellcore priority 4 Short-Long-Short Call Waiting Tone used with this Ring: Call Waiting priority 4 Alert Info string used : Number of States = 7; State1 = ToneOn; State1 Length = 300; State1 Freq1 = 725; State1 Freq2 = 750; State2 = ToneOff; State2 Length = 200; State3 = ToneOn; State3 Length = 1000; State3 Freq1 = 725; State3 Freq2 = 750; State4 = ToneOff; State4 Length = 200; State5 = ToneOn; State5 Length = 300; State5 Freq1 = 725; State5 Freq2 = 750; State6 = ToneOff; State6 Length = 4000; State7 = ToneRepeat; State7 Reps = 0; Bellcore priority 5 - Short - "Ringsplash" Call Waiting Tone used with this Ring: Call Waiting priority 4 Alert Info string used : Number of States = 1; State1 = ToneOn; State1 Length = 500; State1 Freq1 = 725; State1 Freq2 = 750; Call Waiting tones from Bellcore specs Bellcore Call Waiting priority 1 - standard Call Waiting cycle Number of States = 1; State1 = ToneOn; State1 Length = 300; State1 Freq1 = 440; State1 Freq2 = 440; Bellcore Call Waiting priority 2 - Call Waiting Priority 2 cycle Number of States = 3; State1 = ToneOn; State1 Length = 100; State1 Freq1 = 440; State1 Freq2 = 440; State2 = ToneOff; State2 Length = 100; State3 = ToneOn; State3 Length = 100; State3 req1 = 440; State3 Freq2 = 440; Bellcore Call Waiting priority 3 - Call Waiting Priority 3 cycle Number of States = 5; State1 = ToneOn; State1 Length = 100; State1 Freq1 = 440; State1 Freq2 = 440; State2 = ToneOff; State2 Length = 100; State3 = ToneOn; State3 Length = 100; State3 Freq1 = 440; State3 Freq2 = 440; State4 = ToneOff; State4 Length = 100; State5 = ToneOn; State5 Length = 100; State5 Freq1 = 440; State5 Freq2 = 440; Bellcore Call Waiting priority 4 - Call Waiting Priority 4 cycle Number of States = 5; State1 = ToneOn; State1 Length = 100; State1 Freq1 = 440; State1 Freq2 = 440; State2 State = ToneOff; State2 Length = 100; State3 = ToneOn; State3 Length = 300; State3 Freq1 = 440; State3 Freq2 = 440; State4 = ToneOff; State4 Length = 100; State5 = ToneOn; State5 Length = 100; State5 Freq1 = 440; State5 Freq2 = 440; ---/snip Can someone tell me how to make * asterisk "send" the relevant signal to a SipTone so it rings differently than the others? Apologies for the long post. Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digi
RE: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Andy Powell > Sent: 25 September 2004 16:27 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway > > > On 25/09/2004 at 14:31 Arik Funke wrote: > > > >Hello together, > > > >I am setting up a communication server which should also act a > >very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 > >MB usb memory stick. What is the ABSOLUTE minimum space requirements for > >~ running asterisk to work as gateway between isdn and lan? 50MB or 1 > >GB?(I would compile, configure, etc. on a separate machine and then copy > >everything to the flash device.) > > > >Cheers, > >Arik > > You could start buy downloading my .iso (29mb bootable ) and use > that as a basisis for your > system. I've already modified it for a CF card based system. > Essentially it depends what sort > of interface to the pstn you want. E1/T1 and analog should work > fine with my cd - but I've not built > it for use with CAPI or the QuadBRI cards... > > you can grab it at http://www.automated.it/asterisk/ > > It's not v1 of * but I am trying to find the time to update to a > newer CVS version, however I will only do that > once I'm happy running that particular version myself... > > > HTH > > Andy > Andy, I would be interested in a CF version too. Please, keep us posted on any progress. Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cheapest SIP Phone
Can I contact you off-list? Please provide email address. Yiannis Costopoulos. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of SeshKanuri > Sent: 22 September 2004 22:41 > To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Shaun > Ewing' > Subject: [Asterisk-Users] Cheapest SIP Phone > > > Folks! > > Our Phones are cheap and they are selling well. We have no complaints so > far. These phones are made by ATCOM, 2nd largest maker of VOIP gear in > China. We are ATCOM's US distributors. > > We want to beat grandstream both at features and price. > > I can sell these industry standard PA1688 Chip enabled phones > with IAX2, yes > I said IAX2 (along with SIP, H323 and MGCP and a few more such protocols > already enabled) at bulk rates to anyone interested in them. > > If you guys are in NJ or NY or nearby areas, I can arrange free demos of > about 100 units at your office locations. > > The link for a pic is here: http://ipphone.eezeephone.com > > Call me at 732-387-4133 > > Seshu Kanuri > Netweb Group, Inc. > Ph:1-732-387-4133 > Fx:1-413-812-3152 > [EMAIL PROTECTED] > www.netwebgroup.com > > "This e-mail message may contain confidential, proprietary or legally > privileged information. It should not be used by anyone who is not the > original intended recipient. If you have erroneously received > this message, > please delete it immediately and notify the sender. The recipient > acknowledges that Netweb Group, Inc. or its subsidiaries and associated > companies, are unable to exercise control or ensure or guarantee the > integrity of/over the contents of the information contained in e-mail > transmissions and further acknowledges that any views expressed in this > message are those of the individual sender and no binding nature of the > message shall be implied or assumed unless the sender does so > expressly with > due authority of Netweb Group, Inc. Before opening any attachments please > check them for viruses and defects." > > - Original Message - > From: "Huddleston, Robert" <[EMAIL PROTECTED]> > To: "'Shaun Ewing'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - > Non-Commercial Discussion'" <[EMAIL PROTECTED]> > Sent: Wednesday, September 22, 2004 8:03 AM > Subject: RE: [Asterisk-Users] SIP Phone > > > > Anyone know where we could get a cheap sip > > phone... We've been playing with an Innomedia MGCP and SIP adapters and > > failing - so thinking that testing with a real phone might be good.. > > > > > > Robert A. Huddleston, KF4BYY > > IT Support Analyst > > Cavalier Telephone LLC. > > (Cell) 804.400.3686 > > [EMAIL PROTECTED] > > > > > > > > -Original Message- > > From: Shaun Ewing [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, September 22, 2004 11:04 AM > > To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial > > Discussion > > Subject: Re: [Asterisk-Users] SIP Phone > > > > On Wed, 22 Sep 2004 16:40:04 +0200, Michael Bielicki > <[EMAIL PROTECTED]> > > wrote: > > > Cisco 7940 :) > > > > I'll concur with that. > > > > The Cisco 7940 and 7960 phones have great speakerphones :) > > > > As for ones to stay away from - the Grandstream BT-100 series. The sound > is > > fine on the local end, but is very low for the remote end (sounds as if > the > > microphone in speaker mode is actually the mic on the handset). > > > > -Shaun > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP <->h.323
Hi, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Linux for Asterisk
Have there been noticed any differences in echo from distro to distro on the very same hardware? I mean install a distro compile and run *, then replace it with another distro on the same box and cards. That could be intersting. Thanks, Yiannis. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: 28 July 2004 14:29 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best Linux for Asterisk On Wed, 28 Jul 2004 09:13:37 -0400, Eric Kirkland <[EMAIL PROTECTED]> wrote: > Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run > Asterisk? I'd like to be able to run the Text To Speech apps and some of > the extended functions of the software (no phone hardware needed, all Voice > over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion > 10 I think?) but I'm having difficulty compiling the TTS stuff. > > I'm just wondering if there's a widely used version that pretty much works > with everything...? I personally use Fedora Core 1 and 2 successfully at home. Gentoo seems to be the most widely agreed upon distribution though. I don't think anyone would slam you for using Asterisk on it. HTH, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Red Alarm after echo on the line
Hi, I have an Asterisk box with two P100X connected on two UK (BT) lines. When I make a call, sometimes, I get too much echo and the system gives me a Red Alert. After that the line is busy. If I disconnect the line from the X100P and I call the line, I get ringing tone. When I reconnect the line on the X100P I get busy tone. For some reason the X100P gets to a state that stays open. Any ideas? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Inbound Issue
Hi, I think that the problem is with the codecs. Search the Wiki and the list archives (through Google) to find what settings in sip.conf you need for Budgetone and Sipura. The settings you need are *allow* and/or *disallow*. Yiannis. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: 26 July 2004 01:52To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P Inbound Issue Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone connected to the sipura Account on Stanaphone.com for eitherbound SIP calls. (I have other SIP accounts as well - all work flawlessly) I have a simple dialplan - an incoming call rings all phones and goes to voicemail if not answered. When I dial '8' followed by a number - the call routes out via Stanaphone fine. No issues. When I call the Stanaphone number - all phones ring as expected, I can answer the call and talk fine. no issues at all. When I dial '9' followed by a number - the call routes out via the POTS line just fine. No issues. However, inbound calls on the POTS line are the issue. When a call comes in, * detects it and starts ringing all of the extensions. However, when I pickup the extension - it gets immediately disconnected. Other SIP extensions keep ringing - and the caller still hears the ring tone. Caller hangs up - SIP extensions keep ringing. Phone I picket up I now return to the hook. * then 'calls me back' ! Does anybody have any idea what's going on? I have put some snippets from the configs below.. Any insight would be very much appreciated! Michael. EXAMPLE FROM: zapata.conf [channels] busydetect=1busycount=7callprogress=yesrelaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yesusecallerid=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1pickupgroup=1-4immediate=nocontext=from-bellsignalling=fxs_kscallerid=asreceivedchannel=1 EXAMPLE FROM: extensions.conf [from-bell]exten => _.,1,Dial(SIP/001&SIP/002&SIP/003,30,t)exten => _.,2,Answerexten => _.,3,Wait(1)exten => _.,4,Voicemail(u099)exten => h,1,Hangup EXAMPLE FROM: sip.conf [002] ; Line 1 on adaptertype=friendusername=002secret= host=dynamiccontext=extensionsmailbox=099incominglimlit=2canreinvite=no
[Asterisk-Users] rxgain - txgain values
Hi, I know that this issue has been discused guite a lot, but I haven't managed to get a definite answer. Is those two values supposed to be floats (e.g. 3.5) or integers with the percent symbol (e.g. 20%)? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing X100P
The extension of an incoming call through the X100P is s. So,
[incoming]
exten => s,1,Answer
exten => s,2,Dial(SIP/200)
exten => s,3,Hangup
[outgoing]
exten => _9.,1,Dial(ZAP/g1/${EXTEN,1})
You need to dial 9 from your SIP phone to get an outside line and then the
number you wish to dial.
g1 stands for "group 1". Add this into your zapata.conf under the X100P.
Yiannis.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wiley E.
Siler
Sent: 21 July 2004 08:30
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Installing X100P
That did it. I have the wcfxo running and channeled. Now I just have
to beat my dial pan. I can dial internally to all my SIPs but outbound
and inbound off the X100P are still not running. Do I just do this...
Define [incoming] in extensions
[incoming]
exten => 1234567,1,Dial(SIP/2000) ; 1234567 = a local incoming call
number?
exten => 1234567,2,Congestion
Is this correct?
Thanks for the help!
Wiley
-Original Message-
From: Seth Remington [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Installing X100P
Install the kernel-source RPM off of the RH9 CD.
-Seth
On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote:
> The error I receive when I run make
>
> Thanks,
> Wiley
>
>
> -Original Message-
> From: Wiley E. Siler
> Sent: Tuesday, July 20, 2004 4:12 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Installing X100P
>
> Could this have to do with the fact that I do not have a copy of the
> redhat source code in the palce specified immediately at the top of
> Makefile? The writer makes reference to Redhat breaking stuff and
> that the headers... Here is is...
>
> # Okay, the people at RedHat have to break everything they can
> possibly even attempt to.
> # So, we have to look in /usr/src/linux-2.4/include for header files
> given their brain dead # crappy installation. (Mind you, I'm a RedHat
> user myself, so I suppose I'm just as # stupid as they are). Everyone
> else who is mildly sane of course links /usr/include/linux # to their
> working kernel source directory, the way God himself does, of course #
> (assuming He's running Linux -- which we all know He must).
>
>
> Well, I do not have a copy of those src files lcoated there. I
> installed from Redha 9.0 cds. Do I need to get a copy of the linux
> kernal source before I compile the zaptel stuff?
>
> Thanks,
> Wiley
>
>
> -Original Message-
> From: Seth Remington [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 20, 2004 2:09 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Installing X100P
>
> You have to compile and install zaptel *before* asterisk for that to
> work. You don't have to change your version, just "make install" in
> zaptel source directory and then "make clean" & "make install" in
> asterisk source directory.
>
> -Seth
>
> On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote:
> > I attempted to install an X100P card but it was not correctly
> > recognized by my Redhat 9 install. I had a test install running
> > without any cards which was working great minus the outward dialing
> > since no cards existed. Now that I have a card, I want to add it to
> > the system. Do I have to scratch the whole current install in order
> > to get the X100P running on my system or is there a way to get it
> > installed as is? I really do not want to change my version of
> > Asterisk since it is running well at this point. Is it possible to
> > just update and add the card?
> >
> > Thanks,
> > Wiley
> >
> --
> Seth Remington
> SaberLogic, LLC
> 661-B Weber Drive
> Wadsworth, Ohio 44281
> Phone: (330)335-6442
> Fax: (330)336-8559
>
> ___
> Asterisk-Users mailing list
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> Asterisk-Users mailing list
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--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
___
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[Asterisk-Users] Up to date?
Hi, before you start throwing stones to me let me tell you that I am a bit new to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July 2004, as described in Andy Powell's "Getting Started with Asterisk" (http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read about the Asterisk 1.0 RC1, and I would like to download it and install it. Could someone tell me what is the best way to proceed, considering that I already have a configuration that I would not like to loose, and that I would like to have the option to roll-back to the version I already have, if all goes pear-shaped? TIA Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP Phone recommendation
Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? What about soft phones? Any recommendations there (for Windoze and Linux)? Thanks, Yiannis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Display
Hi, I am thinking of using * with IP phones instead of a hardware PBX. The situation is like this. I have 3 different companies having an analogue line for each of them. I want to make sure that when a call comes in, we have an indication on the IP phone which line the call comes from. Is this possible? Do I need IP phones with 3 lines? Are there any IP Phones with at least 3 lines? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?
Hi, for SIP account you can use this: http://www.freeworldialup.com/ for a UK number try this: http://www.calluk.com the numbers are free. Regards, Yiannis -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Johannes van HulstSent: 14 July 2004 19:42To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Where can i get an UK SIP account with UK number? Can somebody help me with some names of good UK SIP providers? I am looking for a UK number to connect to my asterisk server. Can somebody help me? Regards, Han
RE: [Asterisk-Users] Multiple X100P in Asterisk box?
I can source second hand X100P for a very low price. Since it is for temporary use, I would like to use those. So can I hvae 5 or 6 of them in 1 box? Will Asterisk be able to "see" them as separate entities? Thanks, Yiannis. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Chris Bond > Sent: 27 June 2004 16:11 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Multiple X100P in Asterisk box? > > > You can use the new digium TDM cards with 4x FXO modules you'd only need 2 > then. > > -----Original Message- > From: Yiannis Costopoulos, Web2Net Solutions Ltd. > [mailto:[EMAIL PROTECTED] > Sent: 27 June 2004 16:11 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Multiple X100P in Asterisk box? > > Hi, > > I am the "IT guy" at a small startup based in UK. At the moment we > have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. > Later on we should be changing to ISDN30. > > One of the partners mentioned getting an analogue PBX now, and when > we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen > the website in the past and I know that it can do the job (even > better than > a PBX). > > I need to know, is it possible to have multiple X100P (5 or 6) cards > in the same Asterisk box? What phones should I use in the office? Any > suggestions? > Are soft phones reliable enough on Windows PC's for a small > call-centre/telesales department (5-6 stations)? > > Thanks, > Yiannis. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple X100P in Asterisk box?
Hi, I am the "IT guy" at a small startup based in UK. At the moment we have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we should be changing to ISDN30. One of the partners mentioned getting an analogue PBX now, and when we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen the website in the past and I know that it can do the job (even better than a PBX). I need to know, is it possible to have multiple X100P (5 or 6) cards in the same Asterisk box? What phones should I use in the office? Any suggestions? Are soft phones reliable enough on Windows PC's for a small call-centre/telesales department (5-6 stations)? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
