similar to: outboundproxy timeout or qualify

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of the "outboundproxy=" IP as it should, verified with tcpdump. I've done a

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing

hello, i've a problem with syslinux on an intel sbc2 server board (other boards are ok). versions up to 1.48 work fine but from 1.49 there is a problem. when booting linux the system hangs after the message: Uncompressing Linux... Ok, booting kernel. sometimes there that message isn't displayed and the system just hangs with an empty screen. can you help me? best regards simon

Hi All, I cannot seem to find a resource that will allow me to RAID5 3 x 1tb drives on system install. Can this be done? -jason

Hi, I've used mdadm for years now to manage software raids. The task of using fdisk to first create partitions on a spare drive sitting on a shelf (raid 0 were my 1st of 2 drives failed) is kind of bugging me now. After using fdisk to create the same partition layout on the new drive as is on the existing drive and then using mdadm to finish every thing up is a little tedious. Any

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Up, Any helps in this would be really appreciated. I stuck with this. Regards --- On Mon, 6/13/11, saleh [via R] <ml-node+3593743-1399725529-244676@n4.nabble.com> wrote: From: saleh [via R] <ml-node+3593743-1399725529-244676@n4.nabble.com> Subject: Calling R from Java To: "saleh" <s.alhammed@yahoo.com> Date: Monday, June 13, 2011, 12:35 PM Dear Sir/Madam, Sorry

hello i'm saleh please guide me i like that i have a jmail how i have a jmail please invite me thanks bye --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20051016/ec04b372/attachment.htm

-----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

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general

All, I''ve been trying to figure out a strange bug in one of my applications that I think I''ve narrowed down to a problem with rails fixtures. It seems as though the after_create callback is being run twice when I save a record. This is only happening a) in the test environment, and b) when there is a fixture file for that table. The test below only prints "Doing

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

hi im saleh please help me I WANT SOURSE CODE SIMILAR THIS CODE package asl; import java.awt.*; import java.awt.event.* ; import java.applet.*; public class animat1 extends Applet implements Runnable{ int c = 290, d = 300; int x = 20, k = 20, o = 0, m = 10, n = 0, y = 0; Thread t = new Thread(this); Image img1, img2, img3; // AudioClip a1; public void init() {

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the