similar to: Error redirecting an incoming call of a SIP provider to a local extension

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

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 =

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

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,

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Hello, I have found asterisk is dialing a random number when it recieves a call, would anyone know why? The first thing I noticed found peer 4563 (this is a n Xlite Client) Many thanks, Spencer SIP Debugging Enabled spitfire*CLI> <-- SIP read from 82.70.154.145:5060: INVITE sip:448715046363@iptel.tgfslp.dalmany.co.uk SIP/2.0 Max-Forwards: 10 Record-Route:

Sorry it's asterisk-users@lists.digium.com --- harry gaillac <gaillacharry@yahoo.fr> a ?crit : > Luca, > > you may find help here: > > http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/ > http://www.asteriskdocs.org/ http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large > > ask for help to asterisk-users@lists.digium.org > > Regards >

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

FYI, for those of you who aren't on the serusers list. I'd like to hear how others can get this working in small Asterisk settings; I don't really have the time to implement it, but it looks very interesting. JT >To: serusers@iptel.org >From: Adrian Georgescu <ag@ag-projects.com> >Date: Mon, 31 May 2004 23:05:47 +0200 >Subject: [Serusers] CDR mediation for VoIP

Hello *'s, Every time I started my asterisk error shows: chan_sip.c:Got 200 ok onRegister that isn't a register. Failed to authenticate in Register to <sip:adnan.007@iptel.org>;tag=asfeaa71f Maximum retries exceeded on call xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx@192.168.10.189 for seqno 107 (Request) Registration for adnan.007@195.37.77.99 timed out ,trying again.

Hello Group, I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have "PA 1688" chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy