Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
I would like to use the * as VoIP Gateway. Something like that: A user takes off a phone on a Avaya extension and dials for example 8 to reach the CO Port. Then Asterisk answers and sends a dial tone. The user dails a numer and Asterisk is doing the rest! (Sending the call to an SIP or IAX Provider) John Novack <[EMAIL PROTECTED]> wrote: Daniel Kocher wrote: > I would like to connect a Legacy PBX (Avaya IP Office 406) to an > Asterisk Server. I would hardly consider the IP office a "legacy" PBX Unless, that is, you consider anything other than Asterisk "legacy" IP office is current competition for Asterisk, as is Call Manager You really need to define WHAT your goal is here. Provide a Voicemail for your IP office? or what?? John Novack > The Avaya has 3 CO Ports available. I thought buying a TDM30B card > with 3 FXS ports to connect the * to the Avaya CO Ports. > > Is this the right approach? Does any one have experience with such a > configuration? > > Thanks in advance for all recommandations and suggestions > > Daniel Kocher - Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect a legacy PBX to an Asterisk Server
I am planing to connect a legacy PBX (Avaya Ip Office 406) to an Asterisk Server. I want to use the * as VoIP Gateway. The Avaya PBX has 3 CO ports available, so I thought buying a TDM30B with 3 FXS ports and connect then to the Avaya CO ports. Is this possible? Would this be the right way to do it? Any recommendation? Thanks in advance Housi Mueller - Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refreshing DNS lookups
Does there exist any workaround? In the iax.conf file I need to configure a peer with a FQDN instead of the IP because the IP of this domain changes once in a while. "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote: housi mueller wrote: > The "dnsmgr" in Aterisk 1.4.0 seems not to work. I enabled "DNS lookups" > in dnsmgr.conf but after reloading the conf files * never refreshes DNS > lookups. Any ideas how to debug this issue? The DNS manager is not used very much in Asterisk 1.4 at all; don't expect it to provide any benefits at this point. - Have a burning question? Go to Yahoo! Answers and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Refreshing DNS lookups
Hi there The "dnsmgr" in Aterisk 1.4.0 seems not to work. I enabled "DNS lookups" in dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups. Any ideas how to debug this issue? Thanks in advance Housi Mueller - Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
In this case I would need to purchase an E1 card for the Avay PBX an an other for the *. To save costs, I would like to intent the interconnection over the FXO port. Anyone has done this configuration so far? Robert Boardman <[EMAIL PROTECTED]> wrote: Just done this for a client using an E1 Pri card in the avaya box and a sangoma a102, using qsig , works fine, I wouls recommend this to any oneits been up and stable for two months now Regards Robb housi mueller wrote: > The main goal is that any extension from the Avaya PBX can make long > distance calls using the asterisk server as VoIP gateway (using a SIP > Provider). > It would be also great if from a remote IP Phone (in an other > location), a user could use the Asterisk server to dial in and the * > forwards the call to an Avaya extension. > The Avaya has an VCM card an IP Phones (5610) as extensions. First I > thought to connect the * to the Avaya through the ethernet interface > but then I was reading in forums that there are for Avaya third party > IP phone licence needed and that the communication with oh323 is not > stable. > I thought also putting the Asterisk in front of the Avaya. > Telco T1 -> Asterisk <- T1 -> Avaya PBX > This could be a solution for later one. Right know for testing it > would be to expensive. That's why I thought about the Avaya analog > Asterisk FXO interconnection. > Any suggestions..? > > */Thomas Kenyon /* wrote: > > housi mueller wrote: > > I would like to connect an Asterik server to an Avaya IP Office > IP406 > > and use the * as an VoIP Gateway. > > > > The IP Office has two Analog extensions available. I thought > connecting > > this analog extensions to 2 FXO ports in the * to interconnect > the PBXs. > > > What sort of interaction are you after? It may be a better idea to > try > to intercept the line card with asterisk, or if the IP406 has a > VCM card > then to talk to it through the ethernet interface. > > > Is this possible? Does any one have experience with such a > configuration? > > > > Thanks in advance for all recommandations and suggestions.. > > > > Housi Mueller> > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
The main goal is that any extension from the Avaya PBX can make long distance calls using the asterisk server as VoIP gateway (using a SIP Provider). It would be also great if from a remote IP Phone (in an other location), a user could use the Asterisk server to dial in and the * forwards the call to an Avaya extension. The Avaya has an VCM card an IP Phones (5610) as extensions. First I thought to connect the * to the Avaya through the ethernet interface but then I was reading in forums that there are for Avaya third party IP phone licence needed and that the communication with oh323 is not stable. I thought also putting the Asterisk in front of the Avaya. Telco T1 -> Asterisk <- T1 -> Avaya PBX This could be a solution for later one. Right know for testing it would be to expensive. That's why I thought about the Avaya analog Asterisk FXO interconnection. Any suggestions..? Thomas Kenyon <[EMAIL PROTECTED]> wrote: housi mueller wrote: > I would like to connect an Asterik server to an Avaya IP Office IP406 > and use the * as an VoIP Gateway. > > The IP Office has two Analog extensions available. I thought connecting > this analog extensions to 2 FXO ports in the * to interconnect the PBXs. > What sort of interaction are you after? It may be a better idea to try to intercept the line card with asterisk, or if the IP406 has a VCM card then to talk to it through the ethernet interface. > Is this possible? Does any one have experience with such a configuration? > > Thanks in advance for all recommandations and suggestions.. > > Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Avaya IP Office
I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBXs. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please..... need some help
Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out. I have in one location 8 extensions from a Panasonic PBX KX-TD1232 connected to FXO Ports on an MultiVoIP Gateway from Multitech. On the other location I have 8 SP5100 IP-Phones from Micronet as remote extensions. Between the locations I have a VPN. If I call from an IP-Phone to a local extension on the Panasonic PBX or if a local extension from the Panasonic PBX calls to an remote IP-Phone, the voice quality is o.k. If I receive a call from the PSTN to a remote IP-Phone or if a remote IP Phone calls someone over the PSTN, the person on the PSTN cant almost hear anything. The voice volume ist to low. I tried with different Codecs with Echo cancellation or without, with silence suppression or without, on the Gateway as with the IP-Phones etc. nothing helps! I contacted a company specialised in Panasonic PBX and with them we changed the EPROM on the PBX which increases the volume between trunk lines, but with no results. Now I have the Output Gain on the IP-Phones set to the maximum and on the MultiVoIP Gateway I increased the Output Gain 7 db. The Person on the PSTN can hear now better but with calls between local extensions and remote IP-Phones, the volume is now too loud and with noise and echo. Why everything works well with calls made from local PBX extensions to remote IP-Phones but it does not work with calls from or to remote IP-Phones over the PSTN. Where could be the problem of this issue? Thanks for helping in advance... Housi Mueller New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
That is a good argument. But I am not sure yet. Do you know if there are big voice quality differences between the Digital and the Analog card? HousiRobert Webb <[EMAIL PROTECTED]> wrote: On Wed, 15 Feb 2006 08:59:22 -0800 (PST)housi mueller <[EMAIL PROTECTED]>wrote:> Hi there,> > I would like to connect an Aasterisk Server with a >Panasonic PBX (has E1extension).> I only need 4 Lines. So I thought I could use an >Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 >card which is more expensive.> > I dont now which card to take.> > Please tell me what you think about. I appreciate all >suggestions.> > Thanks in advance> > Housi Mueller> > My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box.Spend a little more now and save in the future.Just my $.02Robert Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question
Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B/TDM2401E Card
Hello, The cards you referenced are BRI cards. We are located in México and do not use ISDN. Instead I could install a optional E1 interface to the D-1232 and purchase for example a TE110P card from Digium. But this solution will cost a lot more money and I am limited with my budget. I dont no if it is worth to spend that extra money. I need to find the best cost effective solution. I appreciate all suggestions. Thanks Housi Mueller [EMAIL PROTECTED] wrote: On Monday 13 February 2006 20:43, housi mueller wrote:> Hi there,>> I plan to use Aterisk in our small office. Until now we used a Panasonic> D-1232 Super Hybrid System. The figure is representing the future> ; configuration I where thinking about to have in the office.>> Question 1:> We need only 4 lines and I thought to buy a TDM04B or a TDM2401E card.> There is quite a price difference. Which card would you recommend me to> buy.>> Question 2:> Is such a configuration as shown on the figure with a TDM04B/TDM2401E card > at all realizable?I am not familiar with that model in particular, but I've done some reasurch about D500. I think all of them have BRI interface so you may consider a BRI http://www.junghanns.net/asterisk/page17.htmlor http://www.avm.de/en/Produkte/Server-Produkte/C4/index.js.htmlor http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htmto interconnect Panasonic and your Asterisk.By my opinion Digium cards are more end user/provider oriented which is not your case. Like they say "card that supports FXS and FXO station interfaces for connecting an alog telephones and analog POTS lines through a PC".Let me know how did you do it.benchev___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM04B/TDM2401E Card
Hi there, I plan to use Aterisk in our small office. Until now we used a Panasonic D-1232 SuperHybrid System. The figure is representing the future configuration I where thinking about to have in the office. Question 1: We need only 4 lines and I thought to buy a TDM04B or a TDM2401E card.There is quite a price difference. Which card would you recommend me to buy. Question 2:Is such a configuration as shown on the figure with a TDM04B/TDM2401E card at all realizable? Thanks in adwance Housi Mueller Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk scenario
Do I need special Router and Switch to handel VoIP??El Flynn <[EMAIL PROTECTED]> wrote: housi mueller wrote:> I am new to asterisk and would like to know if a configuration like shown on the picture with asterisk is correct?> > Thank you in advace> > Housi Mueller> Looks good Click here to donate to the Hurricane Katrina relief effort.<> ___ --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
[Asterisk-Users] Asterisk scenario
I am new to asterisk and would like to know if a configuration like shown on the picture with asterisk is correct? Thank you in advace Housi Mueller Click here to donate to the Hurricane Katrina relief effort.<> ___ --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 architecture
I see you mean that in the second Branch I dont need a second Asterisk server, only IP phones which support SIP/IAX2. I dont know jet about rules. But the two Branches are in different city's. So if somebody is calling an area, in branch one which is now located in branch two, this call should be automatically transfered.Il Neofita <[EMAIL PROTECTED]> wrote: The call generate from branch2 can be send to the asterisk in Branch1 with a trunk the same think the call received from branch1 the only thing that is not cleat how you want transfer automatically the call received from the pstn. What rule you want use? On 9/5/05, housi mueller <[EMAIL PROTECTED]> wrote: Why not? How would you solve then the Brench1/Branch2 issue?? a <[EMAIL PROTECTED]> wrote: >From my point of view I do not see any issue with that scenario. On 9/5/05, housi mueller <[EMAIL PROTECTED] > wrote: I am new with asterisk and hope somebody can help me. Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.) Thank you in advace Housi Mueller__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [Asterisk-Users] Asterisk architecture
Why not? How would you solve then the Brench1/Branch2 issue??a <[EMAIL PROTECTED]> wrote: >From my point of view I do not see any issue with that scenario. On 9/5/05, housi mueller <[EMAIL PROTECTED]> wrote: I am new with asterisk and hope somebody can help me. Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.) Thank you in advace Housi Mueller Click here to donate to the Hurricane Katrina relief effort. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews .com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Click here to donate to the Hurricane Katrina relief effort.___ --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
[Asterisk-Users] Asterisk architecture
I am new with asterisk and hope somebody can help me. Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.) Thank you in advace Housi Mueller Click here to donate to the Hurricane Katrina relief effort.<> ___ --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
[Asterisk-Users] Best costs effective solution...
Need help I lost the overview. The situation is the following: I am working in a small office with one Branche office, wich have right now a Telephone configuration like this. Branch1 PSTN --Digital Panasonic SystemPBX 20 Telephones Branch2 PSTN --Analog Panasonic SystemPBX 10 Telephones Now they like to connect the second Branch office over VoIP. Something like this: Branch1 PSTN --Digital Panasonic SystemPBX 20 Telephones | VoIP | Branch2 Analog Panasonic SystemPBX 10 Telephones Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. Which would be the best costs effective solution. Which hardware Routers, Serves, etc. would you recommend. Thank you in adwance Housi Mueller__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
