search for: sipdomain

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

...uxbod,1,Dial(sip/uxbod,20) exten => uxbod,2,VoiceMail(5001@incoming) exten => uxbod,3,Hangup() [outbound-local] exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion() exten => _9NXXXXXX,102,Congestion() [uri] exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) [macro-uridial] exten => s,1,NoOp(Calling remote SIP peer ${ARG1}) exten => s,n,Dial(SIP/${ARG1},120,tr) exten => s,n,Congestion() When I try dialling 912345678 the above...

Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from bob at internal.com to charles at external.com I have added the following lines in extensions.conf exten =>

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <- don't mind L, a quick hack for dtmf not working with sip exten => _1098933X.,11,Hangup exten...

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

...tel.org/ser/) in the "utils/gen_ha1" directory - there is a password generator in there, along with C routines for WWW-Authorize responses. Notes: Replace "123.123.123.123" with the IP address of the remote SIP server that you're trying to REGISTER with. Replace "sipdomain.company.com" with the domain you're using. This may not matter much for some SIP servers, but others are fussy about it. Change the "Expires: " value to whatever you think is useful. It's measured in seconds. Change "John Doe <sip:jdoe@foo.bar.net>" to...

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

...Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=XXXXXXXXX SIPURI=sip:303 at XXXXXXXXXXXXXXXXX CDR Variables: level 1: clid="Ext. 303" <303> level 1: src=303 level 1: dst=XXXXXXXXXX level 1: dcontext=XXXXXXXXXXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009...

...I use outgoing URI-dialing for my sip-phones as suggested in >>> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial >>> >>> The relevant extensions look like this: >>> >>> [dial-uri] >>> exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> >>> [macro-uridial] >>> exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)}) >>> exten => s,n,Set(CALLERID(number)=${CAL...

...lass=3D043373513-29072008><FONT face=3DArial=20 size=3D2></FONT></SPAN>&nbsp;</DIV> <DIV><SPAN class=3D043373513-29072008><FONT face=3DArial size=3D2>On = asterisk's=20 sip.conf&nbsp;I created a user <A=20 href=3D"mailto:conference at sipdomain">conference at sipdomain</A>&nbsp;(sipdo= main =3D=3D=20 server name), that registers to OpenSER when asterisk is=20 started.</FONT></SPAN></DIV> <DIV><SPAN class=3D043373513-29072008><FONT face=3DArial size=3D2>The = user conference=20 refers t...

...L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55xxxx" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya.com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF...

...on'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 the originator rather than the destination, that would be cool, or B) make the inbound gateway use the sip.conf file section belonging to it via the host= line in the sip.conf file without user authentication, or C) some other way I have yet to fathom I'm trying to differentiate between...

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>

...IP URLs. I've found two approaches for doing this in Asterisk: - one is to prepend some numbers at start and catch them - the rest of called string is used for SIP URL - another approach (that I like better) is to use catchall pattern at the end of context _. and then parse string with help of SIPDOMAIN variable. But there is a catch into this one - it only works from sip devices, cause only they populate sipdomain variable with proper info... Is there any better way to do this from all clients regardless of their OS and protocol ? Thanks in advance, regards, Rob.

I use client Sjphone which work fine but i have Sniff a traffic.. - Sjphone send packet with OPTIONS to Asterisk - Asterisk ask with 404 not found This sequence come back often in my log. I don't understand why Sjphone Sens OPTION, and 404 not found.. Thanks for your help

...hich the docs I've found suggest is valid), then I get: 2006-05-12 07:36:16 WARNING[95290]: chan_sip.c:12539 reload_config: Empty context specified at line 43 for domain 'domain.com' and the domain does not appear when I do a sip show domains. It isn't recognised as local, CHECKSIPDOMAIN doesn't do what I want and calls I want are rejected. If I specify autodomain=yes, then the IP address and canonical hostname of the box are added to the domain list, and sip show domains shows them with a context of (default). It would appear that for incoming calls from PSTN gateways at...

...4 seconds before spawning each process. When i input 1, everything works fine (i guess). * records the voicemail (i am sending the contents of a .wav file to asterisk) . Here is the screen capture: *CLI> -- Registered SIP 'gopi' at 192.168.68.15 port 5061 expires 120 == Setting SIPDOMAIN to : 192.168.68.6 -- Executing Dial("SIP/gopi-bddf", "SIP/stest|10|tr") in new stack == Everyone is busy at this time -- Executing Ringing("SIP/gopi-bddf", "") in new stack -- Executing Answer("SIP/gopi-bddf", "") in new stac...

...s best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com' -- Executing [blah at anveo_sip:1] Dial("PJSIP/demo-alice-00000005", "PJSIP/blah at mytrunk") in new stack -- Called PJSIP/blah at mytrunk -- PJSIP/mytrunk-00000006 is ringing -- PJSIP/mytrunk-00000006 is ringing -- PJ...

...e following flags set in the [general] section of my sip.conf [general] nat=yes qualify=yes rtpkeepalive=60 rtptimeout=90 rtpholdtimeout=300 canreinvite=no context=sip_incoming (... among others ...) Following is the relevant portion of my extensions.conf [sip_incoming] exten => _.,1,GotoIf($[${SIPDOMAIN}=mydomain.com]?4) exten => _.,2,Dial(SIP/${EXTEN}@${SIPDOMAIN}) exten => _.,3,HangUp() exten => _.,4,Dial(SIP/${EXTEN}) exten => _.,5,HangUp() exten => h,1,HangUp() Am I doing something wrong? Or is there a bug in Asterisk, wherein, while calling out to non-locally subscribed users,...