similar to: FWD network from Asterisk through NAT

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact info: hank@hanksmith.net Email: Same as MSN. ----- Original Message ----- From: "hank"

- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact info: hank@hanksmith.net Email: Same as MSN. ----- Original Message ----- From: "hank"

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",

hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up

hello how can I set oggenc to encode cds used with abcde to encode ripped tracks in mono? thanks hank <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 - -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hello I have a router that is behind a nat, I want to take asterisk out of nat so I can use it with sip. what would be the best way to go about doing this? I have cable internet and everything is hooked up to a router currently. thanks hank - -----BEGIN PGP SIGNATURE----- Version: PGP 8.1

hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am

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Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

hello where can I get the asterisk prompts that are included in the sample config at? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

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

I have a standard BT home DSL, which means I cannot have a static IP address, therefore i'm forced to use NAT, I subscribe to a DDNS service and have written a VB app which polls the router every 10 seconds and updates the DDNS if appropriate. This is fine but I need to be able to modify my sip.conf (externip = w.x.y.z) and reload sip, does anyone know of a script/app which does an nslookup

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

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

Hi everyone, I would like to attend the conference, but may not be able to make it for sure. Could you put me down as unconfirmed, please. Thanks, Kelly Wilson, M.Sc. Independant University of Calgary Alumnus

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

----- Original Message ----- From: "hank" <hanksmith4@earthlink.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? >I am using asterisk@home 1.0 > my mp3 is called > mp3 > it has nothing before it