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 => cont

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 patte

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 configu

Re: [asterisk-users] Sipura phone does not ring

n 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 <[

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

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 b

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 ot

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.conf for more info  by default incoming calls goes into default context have you checked if registration has occured in sipproxy?check debug messages in asterisk console  2006

[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 know how to check if there is trafic in a channel and otherwise shutdown it? ___ --Bandwidth and Colocation provided by Easynews

[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