similar to: Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?

We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had

Greetings, Somehow my googling skills are not upto mark. Pray can somebody point out resources which works like GnuDIP but listens to Dyndns protocol as one of the proprietary gateways can talk only in dyndns or ddo. Basic scenario: one Private DNS server updating itself using data thrown at it by various known adsl based client (gateway device) using dyndns / ddo protocol. This server will be

Hi out there, I think i've everything set up properly, outbound calls are working fine, but at incoming calls I can't hear anything, but the other one is able to hear me perfectly. I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to my sip-provider using a trunk. Firewall settings on the router are: forward UDP port 5060,5004,10000-20000 to asterisk server

Greetings, I know this mail is slightly OT. But my DNS server and and all other compute nodes (currently around 80) run Centos. Has anybody come across some package script which can be used to listen to request from DynDNS.org clients which can be found commonly in the devices? I have a scenario wherein many devices having Dynamic IP scattered across a geographical region. most have an embedded

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For some reason I now seem unable to control which codec is chosen. The problem happens with outgoing calls to Stanaphone. Even if I chose disallow=all and allow=ulaw as the only codecs it connects with GSM. Has anyone else got problems with these settings? Any suggestions? As I recalled it, such a setup would not establish a call if the ulaw-codec was not offered by the provider. Stanaphone has

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I'm running two servers, one with a fixed IP address and the other with a dynamic address. This is probably a very ignorant question, but what does dyndns do that I could not do myself? -- Timothy Murphy e-mail: gayleard /at/ eircom.net School of Mathematics, Trinity College, Dublin 2, Ireland

Hello I was directed to discuss this issue here. As I understand the issue with using the unquoted variable is that it expand globs unless noglob is set. E.g. root at dy3:/# test="b*" root at dy3:/# a=($test) root at dy3:/# echo ${a[0]} bin It does seem a bit hypothetical that the output of sambatool dns query ... for an A record should contain a glob, but for the sake of robustness it

On Fri, 11 Aug 2023 14:03:01 +0200 Kasper Brandt via samba <samba at lists.samba.org> wrote: > Hello > I was directed to discuss this issue here. As I understand the issue > with using the unquoted variable is that it expand globs unless > noglob is set. E.g. > > root at dy3:/# test="b*" > root at dy3:/# a=($test) > root at dy3:/# echo ${a[0]} > bin

Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its

On Fri, Aug 11, 2023, at 2:58 PM, Rowland Penny via samba wrote: > On Fri, 11 Aug 2023 14:03:01 +0200 > Kasper Brandt via samba <samba at lists.samba.org> wrote: > > > Hello > > I was directed to discuss this issue here. As I understand the issue > > with using the unquoted variable is that it expand globs unless > > noglob is set. E.g. > > > >

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net

Hi, I'm having a problem with my asterisk, trying to connect to a CISCO 2840 IOS12.x ASterisk is behind firewall NATing, when it do the handshaking for RTP, it sends his internal IP instead of sending the external one. How can I tell the asterisk box, to modify that and send the external IP? I tryied with Sip.conf's externip=xxxx and localnet=xxxx, nat=yes Nothing seems to change the

Hi, I''ve a little problem, I hope so.. First a hint, I haven''t a static IP - Adress and so I used a dyndns Provider. In DMZ runs a sftp server. It should accessible from net. My router is forwarding the traffic from port 22 to the machine in DMZ. Now, in basic installation I have rfc1918-dropping configured by net interface. My problem: If rfc1918 dropping is on I

#!/bin/bash # checksetexternip.sh # Author: John Cahill email at johncahill.net # Licence: GPL v3 # Description: script that queries checkip.dyndns.com to find the server's external IP address. Updates asterisk's externip value and does a sip reload if necessary. # Last modified 06/02/2012 is_ip(){ input=$1 octet1=$(echo $input | cut -d "." -f1) octet2=$(echo $input

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

Hello. Thanks for the reply. Yes. In the traffic analyzed, the BYE is sent by the originator of the call, but there is no "human" hangup, but the asterisk one. BYE is sent, received and confirmed. I don't know how I could investigate the reason for this BYE. Em 21/09/2020 17:12, Dovid Bender escreveu: > Is there anything in the Asterisk logs? Which side sends the BYE? Were

All I've suddenly lost incoming audio on my FWD connection. It worked fine up until Wed when all of the sudden my calls would complete but I couldn't hear any audio (I could see the status of the call on the CLI and could see that my call was using bandwidth on the ethernet switch and router). I swear I didn't change any of the configuration or even restart *, but all the sudden