Re: [asterisk-users] How to add include statement into Realtime static
you must use the switch command. I am not sure, but I think you should configure config realtime also, otherwise this command will be in extensions.conf Take a look in voip-info.org 2006/12/12, Tielin Xu [EMAIL PROTECTED]: Hi List: I can not find out an example how to store include = context name statement into Realtime static. Please help me on this one. Thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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] CISCO 2600 - VWIC 1MFT-E1
As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX .115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box. An incoming call in your E1 must much a destination pattern, your only destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example. otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I´ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine. best regards 2006/12/7, FaberK [EMAIL PROTECTED]: http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the dial-peer voice 10 pots, I receive the advise that the number does not exist. Also, I do not find the way to add authentication username asterisk-uername password XX. The story continues... Thanks F. ___ --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] CISCO 2600 - VWIC 1MFT-E1
Hi In dial-peer voice 697617664 voip your must specify into voip dial peer session protocol sipv2 and check if session target sip-server is corect doing a ping to sip-server . I think you must configure it with ipv4:ip_addres or map a host entry with ip host sip-server x.x.x.x in global configuration mode you have forgotten to configure a pots dial peer for your controler. put something like this dial-peer voice 10 pots destination-pattern 0T fax rate disable direct-inward-dial port 1/0:15 and try if you can write authentication username asterisk-uername password XX this last command should allow dial-peer voice 10 to register within asterisk I hope it will help you best regards 2006/12/7, FaberK [EMAIL PROTECTED]: Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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] Attended Transfer
I had problems with featuredigittimeout . It was too short and betwen digit and digit was happened a timeout. modify to featuredigittimeout = 1000 2006/12/5, Arlen Nascimento [EMAIL PROTECTED]: Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set atxfer = * (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento ___ --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 ___ --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] Sipura phone does not ring
Sorry for my delay in answer you.
S0 is how pstn line is identified into spa3000. It means that an incoming
call from S0 will be forwarded to [EMAIL PROTECTED]
yor must configure in sip.conf an account for the sip pstn line and a
context , in extensions.conf the same context with a pattern
or number for dialing.
example
sip.conf
[100]
host=dynamic
type=friend
context=GW-PSTN-ZEL
secret=x
qualify=150
authuser=100
username=GW-PSTN-ZEL
accountcode=ZEL-100
port=5061
disallow=all
allow=ulaw
extensions.conf
[GW-PSTN-ZEL]
exten=s,1,Answer
exten=s,2,NoOp(${CALLERID})
exten=s,3,GotoIfTime(10:00-18:00,mon-fri,*,*?HLaboral,s,1)
exten=s,4,GotoIfTime(09:00-10:00,mon-fri,*,*?DesvioFax,s,1)
exten=s,5,Goto(Cerrado,s,1)
I Have configured the spa 3000 with S0:[EMAIL PROTECTED]
when spa 3000 receive a call, is forwarded to s extension in asterisk, but
before, it is very important that sip user line be registered with asterisk
for making calls.
I hope it will help you
2006/11/29, Larry Alkoff [EMAIL PROTECTED]:
Fran when you say specify the next hop do you mean the S0 line be an
extension in sip.conf or a context in extensions.conf?
Or should the line simply be tacked on to my [default] context?
Larry
Fran Oliveira wrote:
I think it is wrong. You should specify the next hop with some like this
S0:[EMAIL PROTECTED]
2006/11/23, Larry Alkoff [EMAIL PROTECTED]:
Problem: SPA3000 phone does not ring for incoming PSTN call although I
can dial out.
I set up my Sipura with the Voxilla Wizard which is pretty good but
leaves out some important details.
The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -
Dial Plans -
Dial Plan 8 (S0:66610)
Should I put extension [66610] in sip.conf with a context in
extensions.conf that will contain dialing instructions?
Can someone please tell me what the entries under [66610] and the
associated context would look like?
Or just tell me how to handle this - I'm been stuck for some time with
this.
The Wizard was nice enough to give detailed settings for sip.conf and
extensions.conf but nothing about to handle Dial Plan 8 except You'll
need to enter the extension you wish to forward all incoming PSTN calls
to on your Asterisk server. I don't understand how to do that.
Larry
--
Larry Alkoff N2LA - Austin TX
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Sipura phone does not ring
I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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 ___ --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] SIP trunks: order or type
see http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer perhaps it can help you 2006/8/11, Rich Adamson [EMAIL PROTECTED]: Shaun Hofer wrote: ok maybe I can explain my problem better. There two trunks both have the same details except one is type=peer (and only does ulaw) and the other is type friend (and does ulaw/alaw/g729). Incoming calls should be only going into the type=friend trunk, NOT into the type=peer trunk. Both should be able to make out going calls. Yet depending on the order in sip.conf, the type=peer will receive calls. Marco I understand how type works, thats not the problem. It seems Asterisk is sending incoming calls to a trunk that has type=peer. As you clearly pointed out to every one else that this shouldn't be happening.If I recall correctly (and that could be an issue), Olle postedsomething a month or so ago relative to this. I believe he was headingin a direction that essentially did away with this friend, peer, user stuff. I'm thinking that same post talked about which parameters wereused for finding a match, and the ordering of those parameters (eg, IP,username, secret).I didn't save the post, but maybe he'll read this and repost something more accurate then my memory. ;)___--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 ___ --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] Handling inbound and outbound calls passed from a proxy
you must add option insecure=very|yes|no in sip.conf, see http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conffor more info by default incoming calls goes into default context have you checked if registration has occuredin sipproxy?check debug messages in asterisk console 2006/8/9, kjcsb [EMAIL PROTECTED]: I need to handle the following scenarios:1. UA1 -- SIP Proxy -- Asterisk2. UA2 -- SIP Proxy -- Asterisk -- PSTN gateway (SIP) I have configured a trunk to register with the SIP proxy:trunk1register=user1:[EMAIL PROTECTED]/DID1UA1 calls [EMAIL PROTECTED] and the call is recognised as being to DID1. I set up an inbound route for DID1 and route the call as appropriate. That dealswith scenario 1.I then tried to configure another trunk to handle scenario 2:trunk2context=from-internalhost=SIP.Proxy type=peerregister=user2:[EMAIL PROTECTED]A call to PSTN1 from the UA is passed to the SIP proxy which recognises itas PSTN call. The SIP proxy updates the From details and passes the call to Asterisk which (I presume) puts the call into the from-internal context anddials the outbound route appropriately.However that setup messes up scenario 1 which now gives a 404 back to UA1. Ipresume Asterisk is not differentiating between a call made to user1 from UA1 and a call made to PSTN1 from user2. It's just seeing a call fromSIP.Proxy and putting it into the from-internal context.Could anyone advise how I would set up Asterisk to cope with both thesescenarios? I could setup DID2 but I don't know how to pass the call onto the PSTN gateway. I am using AMP/FreePBX but if someone could advise the generalprinciples I would appreciate it.ThanksCameron___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-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
[Asterisk-Users] Fran Oliveira desea chatear
--- Fran Oliveira desea mantener el contacto con usted a través de algunos de los mejores productos que Google ha lanzado recientemente. Si ya utiliza Gmail o Google Talk, visite: http://mail.google.com/mail/b-fc90bb4559-703cfa7c72-d0d0715025b1d0f2 Haga clic en este vínculo para chatear con Fran Oliveira. Si desea obtener Gmail, la cuenta de correo gratuita de Google con más de 2.600 megabytes de almacenamiento, y chatear con Fran Oliveira, visite: http://mail.google.com/mail/a-fc90bb4559-703cfa7c72-2df196b0ca Gmail ofrece: - Protección eficaz contra el spam - Función de búsqueda integrada para localizar mensajes y un útil sistema para organizar los mensajes en conversaciones - No se muestran pop-ups ni anuncios banner no orientados, sólo anuncios de texto e información relevante respecto al contenido de sus mensajes - Funciones de mensajería instantánea en Gmail Todo ello gratuito. Espere. Todavía hay más. También puede disfrutar de Google Talk: http://www.google.com/talk/ Se trata de una pequeña aplicación descargable de Windows* para realizar llamadas gratuitas a sus amigos a través del equipo. Es una herramienta sencilla y fácil de usar, compatible con cualquier altavoz y micrófono instalados en el equipo informático. Gmail y Google Talk todavía se encuentran en fase de prueba. Trabajamos con esmero para mejorar y añadir nuevas funciones a estos servicios, por lo que es posible que solicitemos sus comentarios al respecto de vez en cuando. Gracias por ayudarnos a mejorar nuestros productos. Gracias, El equipo de Google Para obtener más información acerca de Gmail y Google Talk, visite: http://mail.google.com/mail/help/intl/es//mail/help/about.html http://www.google.com/talk/about.html Si al hacer clic en las URL de este mensaje no le funcionan, péguelas en la barra de direcciones del navegador. * ¿No es usuario de Windows? Ningún problema. Podrá disfrutar del servicio Google Talk igualmente desde cualquier plataforma que emplee clientes de terceros (http://www.google.com/talk/otherclients.html). ___ --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] sip channel monitoring
Hi I have checked that when a network conection is lost, sip channels remain actives,and billing time no stop. does any body knowhow to check if there is trafic in a channel and otherwise shutdown it? ___ --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
