similar to: Dialing a SIP Peer without using register strin

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To:

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Hi, I am trying to dial a registered user via his IP:Port mechanism, but problem is that the audio data is not reaching to dialed user. here is the scenario. caller and callee both are registered at asterisk server. asterisk server is on public ip so no port forwarding and natting necessary there. however caller and callee both are behind router and there is port forwarding enabled and nat=yes,

Hi, my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when user and asterisk are on Local LAN. But my asterisk server is on public ip and users are behind nat, and so this method says unknow host when used on public asterisk server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel.

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hey, guys I've been working on whether libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu [root@xx ~]# libvirtd -V libvirtd (libvirt) 4.5.0 [root@xx ~]# qemu-img -V qemu-img version 2.12.0 (qemu-kvm-ev-2.12.0-33.1.el7_7.4) Copyright (c) 2003-2017 Fabrice Bellard and the QEMU Project developers 1. assign $MYSECRET to libvirt secret using the secret-define and

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

yes, actually this scenario is on remote servers. like SIP/XYZ at 119.18.230.20:5060 SIP/XYZ at 202.68.0.90:5678 audio is ok when dialing without using ip & port as below SIP/XYZ but when i dial using below dialstring SIP/XYZ at 202.68.0.90:5678 or SIP/XYZ at 119.18.230.20:5060 then the problem arises hope you got the idea.. Nasir

Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2]

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

Hi guys, I hope you could help me. I am trying to get connected through AMI but something is not working. Both php code and manager.conf were working well in asterisk 1.4 1. Sometimes it gets connected and sometimes it doesn't: == Connect attempt from '192.168.25.241' unable to authenticate == Connect attempt from '192.168.25.241' unable to authenticate == Manager

Ok. I am confused. I now have conflicting answers to my question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B. I would like to connect to iConnectHere, which uses SIP. Has anybody done this before (using similar equipment to what I have listed above)? And if it is not possible, could somebody please explain why. I don't understand why this wouldn't

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,