similar to: how to register * at FWD from behind NAT

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Good day all, I switched my SIP FWD account to IAX and connect my * in IAX. Working great, but I face one problem: I have an iaxtel account and try to call from there (iaxcomm) my FWD Iax # by 17009xxxxxx. It's ringing but no termination on my *. Calling 17009999612 or 17009<SIP FWD#> is working (SIP FWD# being an normal SIP account on FWD). Has someone some idea on what's

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 was considering buying a devkit, bit it looks like it's not longer available. Will there be a replacement devkit? If not, where to buy a reasonable priced channel bank? Which channel bank was included in the devkit and where can it be bought? (btw I live in Holland) Chris

When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process'

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

Hi, I created an iaxtel account, and was given a password containing an "@" character. The directory pages imply that they change the web login password only. How do I reset my IAX password so that it is usable in the iax.conf file? Thanks, Steve

Hi all, I'm trying to build a little voicemail server based on asterisk here, using Asterisk's "Commedian Mail" application. Unfortunately, I'd expect some people to have trouble using the English prompts that come with asterisk. However, I can't imagine I'm the first person who has this problem, and Commedian Mail seems to support multilingual prompts fine, it's

I'm having trouble with incoming calls that start to ring again a few seconds after the local user hangs up. I get "Starting simple switch" on the FXO port, so it's not just a matter of mistakingly initiating a 3-way call on hang up or something. Picking up the ringing telephone will get a CO dialtone. This happens, say, 50% of the time for incoming calls, never for outgoing

Just wondering if there's any significance to the jacks on the TDM cards. They appear to be RJ-45 instead of RJ-11, and I wasn't quite sure if that's something that makes a difference from the user perspective. . . Thx. B.

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote: >> I do not have any zaptel hardware on the Asterisk box, I could not have >> meetme functioning. I did modify the Makefile in zaptel directory on >> line 168 by including ztdummy as one of the modules to compile in. try modprobe ztdummy This works. Should I include this in /etc/asterisk/modules.conf so that it will

Philipp von Klitzing wrote: > You'll need to provide the CODEC that you are using in X-Lite! > The codec used in Xlite is 711uLaw. I guess it is one of the preferred ones other than gsm. And it is of small size. -- David Kwok FWD#/IAXTEL# : 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Subject: [Asterisk-Users] meetme not working Reply-To: asterisk-users@lists.digium.com I am trying to set up meetme functionality but am unsuccessful so far. When I dial the extension, an announcement says, "That is not a valid conference number, please try again." In order for meetme to work you need either zaptel hardware OR you recompile the asterisk directory and include the

Greetings i have been using asterisk with good success with soft phones and I need to purchase some Sip capable phones to test. How would/do vendor specific phones like cisco 7900's and 3Com VCX 3100 series phones work SIP to SIP if they download the protocol from the Vendor call centers? I want to build my own with Asterisk, and use different phones. They say they are RFC 2543

Howdy! After searching around the internet and looking at countless (different) examples of how to setup Asterisk, I've been unable to locate enough to allow my puny brain to comprehend how to achieve my goal. I'm hoping some kind soul will point me in the right direction. Scenario: Multi-user asterisk installation. Each user has several sip and IAX accounts. Each user has at

I am forwarding my calls from my packet8 phone number to my Free World Dialup account using the Packet8 FWD interconnect codes. I have asterisk registered with my FWD account via IAX2 and have also tried with SIP. When a call comes in Asterisk interprets any DTMF tones twice. IE: someone types 1 asterisk thinks they pressed 11. I have tried the echo test and do not hear any keybounce or echo. I

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002