similar to: using asterisk with iptel addreses

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

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 =>

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

Hi all, I'm trying to connect my Asterisk@Home to iptel.org, but the only I get is Allison telling me "circuit busy now, please call again later" or some thing similar. I'm trying make it by AMP and editing sip.conf and extension.conf, and I read all about it in voip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:smilioto@GMAIL.com IM:

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:

Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two

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,

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 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

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

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

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

hello *'s, Astcc not workin what is correct format for defining 1-database 2-brands 3-trunks 4-routes i define all these things but not workin may be i define in wrong format.I have FXO card installed.can anyone implement it and also my sip phone generates very loud noise wat is that i tried several settings but not hear any voice just noise. sip.conf [general] context=from-sip port=5060

Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom

Where is the documentation on stopping addresses of machines on a Lan being visible to scanning s/w when masquerading through a firewall. tnx Richard

Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and reloading, it keeps the same ip open. The point is that i cant open all ips i need at the same

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like