similar to: FWD<->NAT<->*

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

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

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",

This has probably been answered somewhere, but I'm stumped. I have two Zap channels (FXS and FXO), both working fine. I can call from Zap/1 to Zap/2 and reverse. I've also configured SIP channels, both inside and outside of my firewall. Inside can call outside, and outside can call inside. Also, both inside and outside can make and receive calls to/from Zap/1 & Zap/2. What

Thanks for the info so far. I am still trying to asterisk'ize my ML9.2 firewall box and can't get the external SIP registration to work. If I hook up my Sipura directly to the WAN it registers OK. This is the message I get from asterisk: Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_timeout: Registration for '263872@192.246.69.223' timed out, trying again If tried

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

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

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

My asterisk box is behind a NAT firewall. I have friends that are on Earthlink, Vonage, etc. I'd like to make VOIP calls directly to them rather than going through the PSTN. With Earthlink, I can make this work through FWD peeting numbers, but that's sort of a waste of FWD bandwidth. WIth Vonage, it doesn't work. I suspect this is because of the breakage between FWD and Vonage that

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From:

Hello I am trying to register for fwd from * but having problem and unable to solve it. I keep getting this message *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to '<sip:89699@fwd.pulver.com>;tag=as62a7f29b' NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

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

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

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

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

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 =>

Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Asterisk April 17 CVS. - 2 Mediatrix FXS ATA (4 phones) - 2 Grandstream phones. - sip.conf, zaptel.comnf