Re: [asterisk-users] Gateway Eurotech
2015-03-27 10:52 GMT-06:00 Carlos Rojas crt.ro...@gmail.com: I Ricky I have worked with this gateway few years ago, it's good product, they have gateways with PRI connectors and SIP. The quality is good, and it woks good with asterisk or regular PBXs. Hi carlos , thank for your advice, I could ask a favor?, this is the trunk that I have in my asterisk and the gw tells me Unregistered [testsip] context=boss type=friend host=1.1.1.1 # ip gateway port=5060 canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw dtmfmode=rfc2833 in gateway - General - SIP client Name ip port usersecret testsip 1.1.1.1 5060 myboy my123 -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway Eurotech
I Ricky I have worked with this gateway few years ago, it's good product, they have gateways with PRI connectors and SIP. The quality is good, and it woks good with asterisk or regular PBXs. On Thu, Mar 26, 2015 at 11:16 PM, ricky gutierrez xserverli...@gmail.com wrote: Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway Eurotech
Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
Read it (http://the-asterisk-book.com/1.6/minimale-telefonanlage.html), or regret it! jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
On Monday 29 April 2013, James Wystead wrote: This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? Usually it is an expansion card that plugs into a PCI or PCI express slot on the motherboard; so most people would just call it an analogue telephony card (such as a TDM410P, for instance) or an ISDN card (such as a TE410P). One that connects to the mobile networks would be called a GSM card. Analogue telephony cards are further subdivided into two flavours; FXO (which connects to an exchange line) and FXS (which connects to a telephone, and provides the necessary line bias and ringing voltages). Usually a single card will provide for multiple lines, by fitting either FXO or FXS modules as required. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
On Monday 29 April 2013, James Wystead wrote: This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? For 1 - 2 ports they are usually called an ATA (Analog Terminal Adapter). For more than 2 ports they are usually called Media Gateways. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
Guys and gals - these are all excellent answers - I am not being clear, I think. Let me see if I can illustrate it. If you cannot see my diagramme, let me know and I will make a word-type chart. So, the Ip device at the top is a SIP phone Asterisk Server Gateway /IP - This gateway is where the SIP Trunk is - so, a provider like Packet 8 or Broadcomm would have this - this connects directly to the public telephone system (somehow) - a Digium card would not work for me as I am not looking to connect to a dial tone. - Does this make sense? So, the Gateway/IP based - what the hell is that called? I am sure there is such an animal as most of us have configured SIP trunks on Asterisk - so, I'm thinking that this thing that connect to the public phone system is what we see as a SIP trunk - right? So, how the hell do I do that? Probably not that simple. Thanks! Glen [image: Inline image 1] On Tue, Apr 30, 2013 at 9:11 AM, Eric Wieling ewiel...@nyigc.com wrote: On Monday 29 April 2013, James Wystead wrote: This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? For 1 - 2 ports they are usually called an ATA (Analog Terminal Adapter). For more than 2 ports they are usually called Media Gateways. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
Guys and gals - these are all excellent answers - I am not being clear, I think. Let me see if I can illustrate it. If you cannot see my diagramme, let me know and I will make a word-type chart. So, the Ip device at the top is a SIP phone Asterisk Server Gateway /IP * This gateway is where the SIP Trunk is - so, a provider like Packet 8 or Broadcomm would have this * this connects directly to the public telephone system (somehow) * a Digium card would not work for me as I am not looking to connect to a dial tone. * Does this make sense? So, the Gateway/IP based - what the hell is that called? I am sure there is such an animal as most of us have configured SIP trunks on Asterisk - so, I'm thinking that this thing that connect to the public phone system is what we see as a SIP trunk - right? So, how the hell do I do that? Probably not that simple. Thanks! Glen No, it doesn't make sense to me J If you don't need a dial tone, you don't need the PSTN. If you are using Broadcomm, etc., you simply use your Asterisk's system's Ethernet connection. Let's start with your application-what do you want to accomplish? --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
are you talking about sip to pstn? thats called fxo ATA. On Tue, Apr 30, 2013 at 8:59 PM, Don Kelly d...@donkelly.biz wrote: Guys and gals - these are all excellent answers - I am not being clear, I think. ** ** Let me see if I can illustrate it. ** ** If you cannot see my diagramme, let me know and I will make a word-type chart. ** ** So, the Ip device at the top is a SIP phone Asterisk Server Gateway /IP ** ** - This gateway is where the SIP Trunk is - so, a provider like Packet 8 or Broadcomm would have this - this connects directly to the public telephone system (somehow) - a Digium card would not work for me as I am not looking to connect to a dial tone. - Does this make sense? So, the Gateway/IP based - what the hell is that called? I am sure there is such an animal as most of us have configured SIP trunks on Asterisk - so, I'm thinking that this thing that connect to the public phone system is what we see as a SIP trunk - right? ** ** So, how the hell do I do that? Probably not that simple. ** ** Thanks! ** ** Glen ** ** ** ** No, it doesn’t make sense to me J ** ** If you don’t need a “dial tone,” you don’t need the PSTN. ** ** If you are using Broadcomm, etc., you simply use your Asterisk’s system’s Ethernet connection. ** ** Let’s start with your application—what do you want to accomplish? --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway?
This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? So, I guess, not a SIP trunk, but the device that actually IS the SIP trunk. Am I making sense? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
Here are your answers: 1st question: Anything that makes sense. 2nd question: Maybe Please, explain your setup. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway setup
Hi, can anyone help me how to setup a simple gateway for voip phones on elastix system. I dnt no really how it should be connected in reality...? and how to test it . Regards Upendra. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway GSM x Comercio Indevido ?
Boa tarde a todos, Colegas estou a procura de um gateway GSM para ligar ao servidor asterisk de nossa empresa, o objetivo é interligar clientes e parceiros comerciais a nossa central, reduzindo custo das ligações para celular, porem ao ver o regulamento dos planos oferecidos pelas operadoras vi que trata-se de uso indevido a comercialização do serviço bem como a utilização dos chips em equipamentos como GSM Box, Black Box e equipamentos similares. Enfim, minha duvida é: Como então é realizado a venda que vejo em diversos sites de serviços voip de ligação celular que custa até R$ 0,30 centavos (exemplo) ? Qual seria a melhor solução em equipamento, tendo em mente que a idéia seria que o asterisk possa realizar cerca de 40 ligações simultaneamente para celulares da vivo, tim, claro e oi ? Agradeço a quem puder me esclarecer. Cláudio Duarte -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway E1 = Asterisk ?
Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? thanks for your help Olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway E1 = Asterisk ?
- Olivier CALVANO o.calv...@gmail.com wrote: Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? Audiocodes Mediant gateways are top notch. Their Mediant-1000 supports up to 4 interfaces (T1/E1/J1). --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway E1 = Asterisk ?
On Wed, Apr 28, 2010 at 7:58 AM, Tim Nelson tnel...@rockbochs.com wrote: - Olivier CALVANO o.calv...@gmail.com wrote: Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? Audiocodes Mediant gateways are top notch. Their Mediant-1000 supports up to 4 interfaces (T1/E1/J1). +1 for AudioCodes Median 1000. The AudioCodes Median 2000 supports up to 16 T1/E1s if you need more than 4. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway E1 = Asterisk ?
Redfone it's good! On Wed, Apr 28, 2010 at 10:07 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? thanks for your help Olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)412-2352745 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway E1 = Asterisk ?
- Luis Morales faston...@gmail.com wrote: Redfone it's good! Redfone makes a nice gateway(they also have very good support), although it is TDMoE. The OP specifically mentioned they want a gateway which provides SIP connectivity. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Thank you Hatem, I will try it now Thanks VoipCrazy 2008/9/2 hatem moiz [EMAIL PROTECTED]: you can do the following in sip .conf file register = username:[EMAIL PROTECTED] and after that write the configuration for the user: [ user ] username = host = qualify = secret = and so on, do this in the first of sip.conf file Best Regards On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote: Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
you can do the following in sip .conf file register = username:[EMAIL PROTECTED] [EMAIL PROTECTED] and after that write the configuration for the user: [ user ] username = host = qualify = secret = and so on, do this in the first of sip.conf file Best Regards On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote: Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway errors
Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Ciao VoipCrazy, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? SIP locks if it tries to do DNS queries and doesn't get an answer. Try using a local caching DNS server. HTH -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Igor, From asterisk, when internet is down I can ping all extensions. The same occurs in others instalations, when the internet is down, my lical extensions log off from asterisk. VoipCrazy 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway doesn't ring
Hello all, i have a problem on incoming call's from SIP Provider that ist going through the Asterisk to a Grandstream HT502. The first ring is executed on the HT502 propperly, but no more ring will follow. But the call can nevertheless be answered by a phone on the gateway. If i call the same Gateway through a connected second Asterisk the ringing is done well. If a call is coming through the same SIP Provider to another Gateway i.e. Inalp Patton SmartNode SN4552 all is working fine. Is this a problem on the configuration of the Gateway itselfs or on Asterisk for this gateway? Thanks Hans-Peter Straub ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway VIP450FO and VIP 400FO
Hi everyone! I want to know if anyone has the sip gateway VIP-450FO from Planet (www.planet.com.tw). I´m looking for his firmware because I would like to transform my VIP-400FO (H323) in a VIP-450FO (SIP). Does anyone has this firmware to send to me? Thanks, MCelo. ___ --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] Gateway TIMEOUT
HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ 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
RE: [Asterisk-Users] Gateway TIMEOUT
Hello All, Is there any idea please? HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively#133; I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ 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] Gateway crashes when transferring to external lines
I'm using an Asterisk system with a Zultys MX250 as a media gateway for our PSTN. When I configured this originally, I couldn't make or receive any outside calls - the MX250 would actually crash and restart itself. Very bad. As I was troubleshooting, I found a tip in the Asterisk wiki that recommended disabling reinvites for buggy gateway. So, I tried that and it fixed the problem. But now I'm seeing some a similar issue when I receive an outside call to an Asterisk extension, then that extension does a manual transfer to another outside number. The receiving line will ring, but as soon as it is answered the MX250 will crash. Is it possible that Asterisk is attempting a reinvite when I do this, or could it be a completely unrelated problem? Here's the sip.conf entry that I'm using for the MX: [mx250] context=mx250incoming type=friend host=192.168.1.10 dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw - .Dustin Wenz ___ --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] Gateway service under Asterisk
Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application successfully and I'm able to participate as an audio-only participant. What I want to do now is to include more than one teleconference room. When I connect to asterisk from a phone I want * to ask me for the room I want to connect to and for a password which should be read from a database and will be different for each room. Depending on the selected room, * should dial one sip address or another, which are read from a database as well. Please, note that I am not talking about Meetme rooms (although I don't know if I can archive my goal using it). I only want to dial a new SIP agent depending the selected room. How can achieve this? What additional tools will be necessary? Thanks a lot! Eduardo. Eduardo López Martínez [EMAIL PROTECTED] Isabel Operation Center [EMAIL PROTECTED] DIT - Dept. Ing. Sist. Telemáticos Tf: +34 913367366 (446) UPM - Univ. Politecnica de Madrid Fax: +34 913367333 Madrid SPAIN ___ 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] Gateway service under Asterisk
Hi Eduardo, Check the AGI (Asterisk Gateway Interface) scripting: http://home.cogeco.ca/~camstuff/agi.html http://www.voip-info.org/wiki-Asterisk+AGI If you write your own script that: * plays the welcome message * collects digits to select the conference room * redirects the call to the appropriate SIP address I think you will easily be able to solve your problem. Perhaps the list knows about other options, but this is how I guess I would do it. Good luck! --- Eduardo López Martínez [EMAIL PROTECTED] wrote: Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application successfully and I'm able to participate as an audio-only participant. What I want to do now is to include more than one teleconference room. When I connect to asterisk from a phone I want * to ask me for the room I want to connect to and for a password which should be read from a database and will be different for each room. Depending on the selected room, * should dial one sip address or another, which are read from a database as well. Please, note that I am not talking about Meetme rooms (although I don't know if I can archive my goal using it). I only want to dial a new SIP agent depending the selected room. How can achieve this? What additional tools will be necessary? Thanks a lot! Eduardo. Eduardo López Martínez[EMAIL PROTECTED] Isabel Operation Center [EMAIL PROTECTED] DIT - Dept. Ing. Sist. TelemáticosTf: +34 913367366 (446) UPM - Univ. Politecnica de Madrid Fax: +34 913367333 MadridSPAIN ___ 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 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] gateway VoIP h323
Hello, I will answer my own question. The explanation on the h323. conf is not sufficient to set up a good h323 environment. Nevertheless was able to figure out with some offline help how to configure the h323.conf and extensions.conf so it finally worked. Hope to see some h323 documentation like promised. This will be excellent. No more questions .. case closed... Tjapko. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tjapko Smits Sent: Miércoles, 17 de Diciembre de 2003 11:34 p.m. To: [EMAIL PROTECTED] Subject: [Asterisk-Users] gateway VoIP h323 Hello list, In the h323.conf file there is an explanation towards a [det-gw]. The context mentioned here is detroit. context=detroit. Can somebody give me an example what I need to put in this context if det-gw is a VoIP gateway at address 200.200.200.200 and I like to forward all incoming VoiP calls starting with a 9. There is no such a thing in the h323.conf about this context. I am kinda stuck here. Kind regards, Tjapko. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gateway VoIP h323
Hello list, In the h323.conf file there is an explanation towards a [det-gw]. The context mentioned here is detroit. context=detroit. Can somebody give me an example what I need to put in this context if det-gw is a VoIP gateway at address 200.200.200.200 and I like to forward all incoming VoiP calls starting with a 9. There is no such a thing in the h323.conf about this context. I am kinda stuck here. Kind regards, Tjapko. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateway in proxy mode
[For Google to find for others:] I have a Micronet SP5052 FXO gateway which I currently use in Peer-to-peer SIP mode with my asterisk box. I have been having an intermittant problem with the Micronet cutting off the incoming audio stream (i.e. PSTN to VoIP) so that the user on the PSTN side can hear me, but I cant hear them. I have talked to Micronet about this, and they have said it is a known bug in the SIP Peer-to-peer in their firmware and that they are currently working on a fix. [Help for me] They have said that in the meantime I shoudl use the gateway either in proxy mode or use H323. I would prefer to stick with SIP, so I was wondering, what do I need to do to my asterisk box to allow the micronet to use it as a proxy?
