similar to: FW: clients unable to auth

Hi All, Thank you guys - your advice was spot on. I will now reach out earlier and not struggle with issues like this for 2 weeks J Best Regards, From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Wednesday, April 16, 2014 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

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)

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

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 =>

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to

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,

Am using Asterisk 11.2 in one location and 11.1 in another location. when I trunk between two servers, the status is unreachable. But with different server with 11.2 and 11.2 it works fine. I tried both IAX and SIP. the trunk in sip.conf what i have is, [serverb] type=friend username=serverb secret=serverb host=10.10.10.5 port=5060 context=default insecure=port,invite dtmfmode=rfc2833

Hi, I am using Asterisk 11.5.1. As far as I understood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls (http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place calls though with this config? sip.conf ... [thorsten] type=peer host=dynamic context=my_context nat=force_rport,comedia secret=... dtmfmode=rfc2833 disallow=all

I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia

First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: <[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia" should have returned the public IP the call arrived on, but it

Asterisk Project Security Advisory - AST-2011-013 Product Asterisk Summary Possible remote enumeration of SIP endpoints with differing NAT settings Nature of Advisory Unauthorized data disclosure Susceptibility Remote

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

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

On 2018-09-09 10:27, Antony Stone wrote: <snip > 1. Try removing one of the two commas. > > 2. Take a copy of your dialplan, and then strip out *everything* except > the > one context and the one number you want to match: > > [0705680837] > exten => 31705680837,1,NooP( Incoming 31705680837 on CC) > same => n,Answer(); > same =>

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this