similar to: Problems making SIP URL outgoing dial

I have set up an asterisk server and can successfully call between extensions using SIP. i wish to be able to call other sip users using URLs such as sip:user@sipdomain.com and have no idea how this works... every time i try it (using X-Lite soft phone), i just get a 404: not found error. Any clues? Cheers Dan -- Dan Goscomb <dang@cashcade.co.uk>

Hello, I'm pretty new to asterisk (only been fighting with it on and off for about the last month), so please go easy. I've been wrestling with the documentation, forum posts, google, and my lack of telephony and VOIP knowledge, trying to get my setup to work. My current problem has me stumped saying, "There's gotta be a cleaner way to do this." Please show me the light

Some of you have unusual SIP configurations, and this SIP perl script may be useful to get remote devices registering with your Asterisk or other SIP server. Most Cisco routers, as an example, are too stupid to REGISTER, so this script would be required to dynamically register them with a remote server. This may not be 100% applicable to Asterisk, since static registrations are possible,

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

Hi I'm struggling with setting up SIP domains. If I specify a domain and a context in [general], that context overrides any set in type=peer blocks elsewhere. This results in incoming calls from PSTN gateways I use arriving in the wrong context. If I don't specify a context (which the docs I've found suggest is valid), then I get: 2006-05-12 07:36:16 WARNING[95290]:

How do I prepend all url''s in my application with a specific string so instead of generating URL''s of the form /:controller/:action/:id I would get /myString/:controller/:action/:id? I know I can use routes.rb and url_for() to accomplish a similar task on an individual controller basis but I don''t see a way to easily prepend all urls with a string. Thanks, John --

Hi All, I''m looking to prepend a string to all (RHEL based) PE3 console URLs. I''m trying to proxy the console thru an Oracle web server (Apache under the hood), and for security we need to make all URLs easily identifiable as having originated from Puppet in the web log files. A PCI requirement I believe. So, the question is how to change https://console.puppet.net into

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

Stefan Gofferje wrote: > Hi folks, > > I would like to have kinda catchall function for incoming sip > connections. A channel, where everybody could connect to by dialling the > url e.g. sip://guest@<myserver>, like the [guest] section in iax.conf. > I have played around a bit but any attempt to dial any > extension@<myserver> without prior registration leads to a

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Dear All, I have a tiny dial plan like: [testing] exten => 454,s,Ringing() exten => 454,n,Wait(4) exten => 454,n,Dial(SIP/slee@192.168.45.183:5605,10) exten => 454,n,Hangup This connects fine when I dial 454 from any extension in my system, but there is never any audio? Where can I start to look for debugging this? It's all internal so no NAT problems? Thanks, Gavin.

if the dns has _sip._tcp.my.dom. SRV 0 0 5060 asterisk.dom.ain. _sip._udp.my.dom. SRV 0 0 5060 asterisk.dom.ain. where asterisk.dom.ain. is an A RR of the asterisk pbx. how does a call to sip://user@my.dom come in to asterisk so i can route it? do i just put in sip.conf [username] context = from-url-username and extensions.com [from-url-username] exten =>

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 - can I do this via the polycom config? I can't find anything in the docs. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just discovered that the SIP channel has undergone some major improvements. I'm now able to dial any SIP URL with dial, couldn't get it to work earlier, all domains had to be defined in SIP.conf. This, in addition to the SIPDOMAIN variable, makes the SIP channel even more useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good afternoon everybody. I first would like you to excuse me for my english. I have an issue with a SIP call ID which is not changed in the call configuration described bellow : I have an Asterisk Server A using only SIP protocol. Behind A there are 2 distant clients (using softphone X-lite) C1 and C2 and a proxy server OpenSIPS (ex OpenSER) P. The idea is that when C1 want to call C2,