search for: iptel

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 => _3.,3,Dial(SIP/${EXTEN:1}@...

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 ba...

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???...

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 discovering SRV records, and does not for iptel. It looks like a bug. All modules a...

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 => _3.,3,Dial(SIP/${EXTEN:1}...

...quot;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: Message from iptel.org SIP admin > >At 11:42 PM 2/23/2003, John Todd wrote: >>While I understand SIP to a "reasonable" degree, I don't see what >>the problem is below with the >ACK message. Should it not be >>sending an ACK at all...

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...

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,...

...TN, 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 Message ----- From: braincrew.com To: serusers@iptel.org ; asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 5:00 AM Subject: [Serusers] ser+asterisk - security Hi there, I'm using ser and =sterisk together. Asterisk for voice mail etc and ser for registration of the =sers usig database. I can restrict =orwarding calls...

...). Getting IAX to work with IAXTEL wasn't a problem, but I'm still fighting with inbound/outbound VoIP "trunks" with IAX2 or anything SIP. I can call 1700NXXNXXX IAXTEL numbers, and anything gatewayed from that network (ie, FWD 170099XXXXX gatewayed numbers work). To get FWD, IPTel, or SIPPhone to work, I've deduced I need to do something like the following after reading various online email archives (please correct me if I'm wrong): sip.conf: [general] register => XXXXX:password@fwd.pulver.com/1000 register => 1747XXXXXXX:password@proxy01.sipphone.co...

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...

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 tried [iptel_in] for incoming calls, but commented it out again. My extensions.conf context is called sip-in. I didn't f...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable to create channel of type '...

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 registrations now, but no calls. get a registration effort every 225secs and...

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

...----- Original Message ----- From: "Walter Carter" <carterw@corp.earthlink.net> To: "'Joao Pereira'" <joao.pereira@fccn.pt>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>; <serusers@iptel.org> Sent: Friday, January 07, 2005 3:17 PM Subject: RE: [Serusers] softphones Try Xten: http://www.xten.com/index.php?menu=products&smenu=xlite Regards, WSC -----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@iptel.org] On Behalf Of Joao Pereira Sent: F...

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 noise , i tried several settings of DTMF but all in vein i enabled/disabled DTMF settings but...

..., would i then be > able to provide prepaid service? > > soryy for asking too much, i'd just like to really understand it. Thank > You in advanced. > > Regards, > Nhadie > _______________________________________________ > Serusers mailing list > Serusers at lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

...access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan to install Asteriks on that server - I plan to install a SIP-proxy,registrar on the same server (I've been looking at iptel's SER) - I plan to use the Budgetone SIP phones - I plan to have a public (static) IP address All this to have my own little phone company for me and my family/friends as we are spread over Europe (high international phone costs!). Calling eachother on our SIP phones and also being able to use...

It's been down the last 5 hours at least. Anyone know what the problem is, or when it will be back up?