similar to: Excternip and FWD

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, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] ..... register => 688426:xxxxxxxx@fwd.pulver.com/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426

<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 I have had a TDM400p for a year now, recently I have been having problems with 'static' noise completely drowning out a call ( inbound only not on outbound calls)., has anyone else experience this and able to offer a solution. Also on inbound calls for the first 1/2 second or so I get a 'ray' gun sound and a bit of static then the call settles down and is usable any advice

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

Hi, just to repeat my previous post (and trying to find a solution): Setup is * behind NAT. I can use FWD (time service, echo server) without problems when I add this to sip.conf: externip=a.b.c.d ; a.b.c.d is the IP of the router (Linux/Nat) outside_addr=a.b.c.d My ICH however now responds with: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp("SIP/113-6d2e", "") in new stack -- Executing Goto("SIP/113-6d2e",

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Hi List I'm trying to update to the lastest kernel but I have a dirver that is not inculded in the distrubution, and I had to use the driver disk when installing centos 4.4 in the first place, The driver megasr .ko works fine with the installed kernel but I cannot find on for the updated kernel, any adive would be appreciated. without the updated driver there is a kernel panic on boot due to

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

Hi I have a problem with a new DECT phone I have bought The key pad works like a Mobile phone where you dial first then pick up the line, but it seems to dail too fast or spuriously, ie 012826736464 show on thew Asterisk console as 0012282677, could any one offer advice how to fix? Also when doing a ZAP bridge to this phone from an outside line the call is very echoy, but not an internal

Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb