similar to: SIP Proxy Issues

I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers.

-----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

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

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, I'm trying to obtain confidence intervals for auto and cross correlation estimates. I've adapted code made available by Stock and Watson that uses the Bartlett Kernel and the delta method. In R it runs really, really slow because of the loops it uses and I have 9 series that I'd like to examine (81 total combinations). It was easy enough to replace one of the while loops with a

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

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

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, Thank for all your reply, here the details of the product : https://www.wallix.com/en/access-manager/ ? Customizable admin portal: Fully customize the design of your administrative portal. Determine how it classifies files, and how files are transferred between workstations and targeted Windows sources. Plus, quickly communicate with different target Bastions via the encrypted https

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.

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' --

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use

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

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 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

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really