similar to: problem connecting to icallglobe

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options

Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is

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"

Hi everybody, I am almost there on that one :-) Transfering a DID from SER to Asterisk 1.2.6, but I get 403 forbidden. I tried this example but without success and I also looked at last year's posts.. http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/#3.3 SER is the public access and is on a separate box. The URI is sip:551130256898@sip.provider.com and the Asterisk is correctly

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden

Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

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

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello All, I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be: A bad CVS release - I will try to download and build from a new one Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header. I am

I concur with Jose. The Atmel AVR series packs a lot of bang for the buck. They also come in a 3.3v low power version for use in battery powered systems. Gene -----Original Message----- From: Leo Ann Boon [mailto:leo@innovax.com.sg] Sent: Tuesday, August 19, 2003 7:21 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Open source IP phone, maybe? Ubicom's Scenix IP2K.

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

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,

Hi there, I'm having a hard time getting outbound calling to my SIP-->PSTN gateway. I continuasly get the following result in my log files: Jun 5 10:07:50 WARNING[1568]: No such host: t2y Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP' Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time I make the following context in my

I am having a problem trying to use originate from the CLI on Asterisk 1.8.2.3. The SIP peer is defined correctly and it works if I dial using my IP phone. When I try to dial from the CLI I get this message: pbxoficina*CLI> originate SIP/protel-out/0445540881644 application playback tt-monkeys [Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048 handle_response_invite: Received response:

I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received