similar to: SIP setup

Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

If receive a call on the IP Dialog SipTone II, and the other end hangs up first, the siptone immediately enters into the congestion tone. If I initiate the call from the siptone and the other end hangs up first, same thing -- congestion. The same thing happens if we make calls from the analog phones attached to the Mediatrix 1102. This does not happen on our Snom 200 phones, which have

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

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)

We have a Mediatrix 1102 hooked into the network. Both of the attached analog phones and all of their features work, but in the CLI we keep getting "-- Got SIP response 481 "Transaction Does Not Exist" back from XXX.XXX.XXX.XXX " (Where XXX is the IP address of the Mediatrix ) every few minutes. I have changed most of the settings in the sip.conf multiple times and have done

The X-lite client is installed behind an ADSL router and it seems the ISP also has a firewall supposedly protecting their ADSL customers blocking some netbios ports and non-critical. I can't make outbound calls but I can receive but before then I had to use the following in the account section in SIP.conf [2222] ... type=friend secret=xxxx host=dynamic ;dtmfmode=inband ;

Hey All, We are looking to expand our usage of Asterisk and I am trying to make as much of the configuration dynamic as I possibly can. The only part that I'm having problems with is sip.conf. I can get asterisk to register each extension with our local SER SIP proxy dynamically by using the "sipfriends" table in the database, but I'm having trouble with the message waiting

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

Hello, I'm attempting to get Asterisk running for the first time in my company. As I've never used it before, I am creating a small testbed with which to learn Asterisk and get the kinks worked out before attempting to roll it out. I have * compiled and running, and built the sample config files as suggested by the Wiki. I got my Snom 190 configured to use DHCP, and have created an

Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send

I bought a MV-372 for 2 SIM cards as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: --------- Inet --> VoIP provider ---> POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |----------> SNOM190 phones | ----------> SIP-GSM-module ---> SIM

When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 <109> callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but