similar to: calls dropped after 20 seconds in a non NAT situation

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is

Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi all, I am using * as a PBX for a Broadvoice VoIP account. It had been working well since about last November, although not perfectly (similar disconnection problems, although I am pretty sure it had to do with my PPPoE setup, but I think these issues were resolved). As of a few weeks ago, though, I started having serious problems. Basically, I can start up * and connect to Broadvoice and

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip

Hello everybody, I'm experiencing a strange problem with my Asterisk. I hope you can help: Asterisk is running at my company behind NAT. Ports 5060 and 10000-20000 are being forwarded to it. I have put the router's external IP-address into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN 7050 which is registered with the Asterisk at my company. When I try to call

I had my asterisk server working fine with FWD as a SIP provider, so I now added a second SIP provider (voctel). The addition to my sip.conf file is almost identical to FWD, however, asterisk now generates lots of debug messages for some strange reason! In particular, the line "##### Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my log below). If I comment out

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. I can

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission

<MOD NOTE:Please kill/bounce my other email, it was accidental.> I just pulled down the newest CVS and recompiled. FWD (free world dialup) works now from *, and I AM behind a NAT. I've nearly given up on the xten lite, iaxcomm sounds better. I'll be trying the other win app thats up-and-coming on the list later. It seems to have broken iptel, but that's not as important to

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

Hi all, Seriously, I've tried to read everything I could find (& search for) on voip-info.org and other sites about this problem, but have been unsuccesful. Equipment: xten lite X100P Whitebox linux running Asterisk / AMP D-Link DI-804HV (VPN router) I have installed another DI-804HV at a second location and created a tunnel. For the computers behind that unit, everything works fine

Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

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

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability

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