similar to: Curious problem with NAT

Have you tried NAT=force_rport ? Ashwin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 11:44 To: Asterisk Users Subject: [asterisk-users] Curious problem with NAT Hi list! Since the internal calls work as expected and I can register my Asterisk on an external

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi Matt, I was a bit concerned on the delay if there might be any when my iax link is down? It would be two dial steps right when my iax link is down. But I?m more than happy to try. Many Thanks, Ashwin. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt Riddell (lists) Sent: 30 May 2015 16:55 To: Asterisk Users Mailing List -

Many Thanks Carlos, I was hoping to check whether the remote server is available before I issue the dial in my dial plan. Is there a better way to do it in asterisk without using unix commands? Many Thanks, Ashwin On 5/30/15, 2:06 AM, "Carlos Chavez" <cursor at telecomabmex.com> wrote: >On 5/29/15 1:16 PM, Ashwin Surendran wrote: >>> Hi, >> I have multiple

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > > > What settings have you got for directmedia? > > > > Could you try > > > > nat=force_rport,comedia > > directmedia=no > > Tried. Peer always unreachable, call not possible... :( > >

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,

What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no -Ashwin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 12:04 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Curious problem with NAT

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 =>

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

>Hi, I have multiple Asterisk servers in various parts of the world all connected using dedicated VPN?s. Each of these servers have iax and dahdi TRUNK configured on them. Occasionally the VPN?s fail. What I want to be able to do is on my dial plan, use IAX if the asterisk server can reach the remote server using the internet OR, use TRUNK only if it can?t use IAX. Any ideas on how this

Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME

-----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 again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

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

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Hi, Is there a way in R to get the string which matches the expression, where the expression is a substring of the parent string. Lets say, I have $i <- "transcript:ENST0000112334 pid:ENSP000012345" What I need is the string "pid:ENSP000012345" from $i using the query "ENSP". Appreciate your comments. Praveen Surendran School of Medicine and