search for: 200&sip

I'm trying to set a variable in one context and use it in another: [c1] MY_GROUP = SIP/200&SIP/201 [c2] include c1 exten s,1,NoOp(${MY_GROUP}) The noop prints out a blank string. In [c1], I've tried exten s,1,SetVar(MY_GROUP=SIP/200&SIP/201) and exten s,1,SetVar(_MY_GROUP=SIP/200&SIP/201) and exten s,1,SetVar(__MY_GROUP=SIP/200&SIP/201) and exten s,1,SetGloba...

...ea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see "NO ANSWER" and "ANSWERED" - what would be there if voicemail "answers"? Asterisk: 1.2.23 [macro-ext-group-home] ; ${ARG1} - Virtual Extension (e.g. 2005) exten => s,1,ExecIF($["${RECORDSIP}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CALLERID( num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV) exten => s,2,Dial(SIP/2${ARG1:-2}&SIP/4${ARG1:-2}&SIP/6${ARG1:-2},${OFFICE_TIMEOU T},rw) exten => s,3,Voicemail(u${ARG1}) exten => s,103,...

...hem whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten => s,1,SetCIDName(Company 1) exten => s,2,Dial(SIP/200&SIP/201&etc.,30) exten => s,3,Voicemail(su200) Each company is able to see on the LCD on their SIP phones whether the call is for them or the folks up/downstairs. What I'd like to do is implement a delayed ringing strategy - i.e. if the call comes in for Company 1, only their SIP p...

Hi all, I'm trying to get a better understanding of the 'Hint' priority for use with Snom phones, etc. >From the Wiki, I understand that the following will work: exten => 200,hint,sip/200 exten => 200,1,Dial(sip/200) exten => 201,hint,sip/300 exten => 201,1,Dial(sip/201) When exactly is 'Hint' executed? After someone dials '200' or '201'? Or does the 'hint' priority apply it's magic at dialplan load time? Suppose I w...

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context '...

Hi there, I have a Polycom phone that has two extensions registered to it, let's say 200 & 201. Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy when either of 200 or 201 are in use? Reason is that my Polycom phones will show the presence info via BLF red light, but I would have to have 2 separate entries in the monitoring phone (one for each extension...

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings and I can press 1 to connect the cal...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4 w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've got the grandstream and the GXP-2000 I would like to ring both. Using macro-exten-vm and dialparties.agi Macro(exten-vm,200,200-202) the caller is sent to the unavailable voicemail but if I use Dial(SIP/200&SIP/202) then it works fine (except for calling features). Does an...

...st?!?! It seems to work for a little while after a reboot then dies a death. The extension just gets dead air, and the error message is on the CLI, it always comes up after a call, and seemingly randomly inbetween. Any suggestions where to look? extensions.conf has exten => _[123],3,Dial(SIP/200&SIP/202&SIP/210&SIP/211&SIP/212|40|m) and exten => 7,1,musiconhold() both do the same thing. musiconhold.conf has [classes] default => mp3:/var/lib/asterisk/mohmp3,-z ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => quietmp3:/var/lib/asterisk/mohmp3,-z Thanks Mark

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not wor...

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout before dialing num...

Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik

...YE, REFER Contact: <sip:s@10.0.0.245> Content-Length: 0 to 69.1.236.33:5060 We're at 10.0.0.245 port 11458 Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2 Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164 Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr> Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=o...

...YE, REFER Contact: <sip:s@10.0.0.245> Content-Length: 0 to 69.1.236.33:5060 We're at 10.0.0.245 port 11458 Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2 Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164 Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr> Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=o...

...YE, REFER Contact: <sip:s@10.0.0.245> Content-Length: 0 to 69.1.236.33:5060 We're at 10.0.0.245 port 11458 Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2 Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164 Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr> Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=o...

...omain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 the Dialplan [inbound] exten => _X.,1,Answer() exten => _X.,n,Set(DB(lastcaller/number)=${CALLERID(num)}) exten => _X.,n,GotoIf(${BLACKLIST()}?black,1) exten => _X.,n,Wait(2) exten => _X.,n,Dial(SIP/200&SIP/201,60,tToxX) exten => _X.,n,Goto(ausser-zeit,_X.,3) exten => _X.,n,Hangup() exten => fax,1,NoOp(**** FAX DETECTED ****) exten => fax,n,Goto(fax-rx,receive,1) [fax-rx] exten => receive,1,NoOp(**** FAX RECEIVE ****) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBA...