Re: [Asterisk-Users] simple question...
On Thu, 2004-01-22 at 21:55, Jess Magnaye wrote: > it just came to my mind, and i haven't done any researches yet if > somebody tried this one with asterisk.. :) well just in case somebody > or someone on the list aware, i appreciate any advise. > > in telco world, there's like 64kbps per channel and voice can be > carried on a 16kbps channel. is it possible to configure asterisk to > make 4 extensions (ATAs example), to call out using single FXO port at > the same time? if that is possible, then is it also possible to make > t1-pri to be capable of transmitting 4x23ch simultaneous calls..? Your problem is needing to have the same software on both sides, and your 4 into 1 would have to be going to the same destination as the telco route 64k chunks. Of course the question then is if you have the same software at each end, why are you worring about PRI when you can go full IP and not get burned on the D channel. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
No It does not require. Regards, Sanjay. - Original Message - From: "Drew Miller" <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Simple Question Does AMD (answering machine detect) need ztdummy or some other timer to function properly? -- Drew Miller Iowa Democratic Party Information Technology Director Office: (515) 974-1682 Cell: (515) 451-4509 AIM: ItsDrewMiller MSN: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote: > Does anyone know the purpose of "/n" attached at the end of the dial > command below > > Dial(Local/[EMAIL PROTECTED]/n)< > The 'n' flag tells chan_local not to optimize itself out of the call path. Without the 'n' flag, chan_local will try to remove itself from the call path after the call has been established. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
Rizwan Hisham wrote: > Hi, > Does anyone know the purpose of "/n" attached at the end of the dial > command below > > Dial(Local/[EMAIL PROTECTED]/n )< Yes, and you will too when you read localchannel.txt in your Asterisk source code docs directory. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
Rule of thumb: you first try without the /n; if the new behaviour is different from expected, add the /n :) Just my $0.02 l. On Tue, 01 Apr 2008 17:33:05 +0200, Jared Smith <[EMAIL PROTECTED]> wrote: > On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote: >> Does anyone know the purpose of "/n" attached at the end of the dial >> command below >> >> Dial(Local/[EMAIL PROTECTED]/n)< >> > > The 'n' flag tells chan_local not to optimize itself out of the call > path. Without the 'n' flag, chan_local will try to remove itself from > the call path after the call has been established. > -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple question
Rizwan Hisham wrote: Whats the difference between the following statements in extensions.conf include=>inbound AND #include inbound/*.conf Hi, checkout this page: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf "With the #include statement in extensions.conf, other files are included. This way you can setup a system where extensions.conf is the main file, users.conf contain your local users, services.conf contain various services, like conferencing. This way, the dial plan may be easier to maintain, depending on the size of your setup. The #include statement is not the same as the include statement. The #include statement works in all Asterisk configuration files. " I believe that #include syntax works like a include in programming languages where the file or files listed are included as part of the file that references them. The include => context syntax is for including on context within another. If context A includes context B then calls going into Context A could possibly match extensions in context B. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Simple question
> Whats the difference between the following statements in extensions.conf > include=>inbound > AND > #include inbound/*.conf The first one includes a context the second one includes a file(s). -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/654 - Release Date: 1/27/2007 5:02 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple question
The first include references another context within extensions.conf. Contexts are defined by words in brackets. In your example, there would be a context in extensions.conf that would look like: [inbound] Contexts allow for setting up difference services and difference user capabilities all within the extensions.conf file. The second include is including the contents of multiple *.conf files located in a directory called inbound. JB >>> [EMAIL PROTECTED] 1/27/2007 6:50 AM >>> Whats the difference between the following statements in extensions.conf include=>inbound AND #include inbound/*.conf -- Regards Rizwan Hisham Software Engineer - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Question
On Sun, 14 Nov 2004 16:44:12 -, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Is this quite simple to set up and can I attach asterix to my landline via a > standard modem? > Yes no go to http://www.voip-info.org/wiki-Asterisk and read learn try and read try agin Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] simple question on sip.conf
SW wrote: Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general section of the sip.conf and could land on the PSTN as well. (do not ask my *'s IP address folks, I am not going to run a free PSTN g/w :) So, how do I prevent other than my own proxy to use the general section of the sip.conf file ? As a mater of fact all calls from fwd and iconnect two land on the general section. Hope some one can shead some light. Cheers SW Here is my sip.conf [general] ;calls arrive from sip lands here port=5060 context=default-in Change this to context=forbidden And define "forbidden" context in extensions conf to whatever you like, example exten => s,1,playback(tt-monkeys) exten => s,2,hangup disallow=all allow=ulaw allow=alaw allow=g729 maxexpirey=180 defaultexpirey=160 ;Connect to Free World Dialup (no NAT) register=61358:[EMAIL PROTECTED]/61358 This one does not have a defined context. [fwd.pulver.com] section missing ;Connect to iconnect register=15108688610:[EMAIL PROTECTED]/15108688610 canreinvite=no Same here. [iconnect] ;incoming does not land here, why ? outgoing is fine type=friend secret= username= host=sipauth.deltathree.com dtmfmode=inband ; required by iconnect context=iconnect-in canreinvite=no allow=alaw allow=ulaw allow=g729 [fwd] ;incoming does not land here, why ? outgoing is fine type=friend secret=xxx username=61358 host=fwd.pulver.com context=fwd-in allow=alaw allow=ulaw [vocal] ;used when dialed in from vocal not working type=friend host=ip of vocal server disallow=all allow=g729 allow=ulaw allow=alaw port=5060 canreinvite=no context=vocal-in [vocal-out] ;used to dial out to vocal type=friend host=ip of vocal server allow=g729 allow=ulaw allow=alaw port=5065 canreinvite=no [6300] type=friend username=6300 context=intern ;secret=blah host=dynamic ;defaultip=192.168.254.4 dtmfmode=info nat=1 [6301] type=friend username=6301 host=dynamic dtmfmode=inband context=intern nat=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Olle E. Johansson, [EMAIL PROTECTED] Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net Runbovägen 10, 192 48 Sollentuna, Sweden Phone: +46 8 594 78 810, Fax: +46 8 594 78 820 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simple question on sip.conf
Hi Have you got the context set-up in the sip.conf to say which extension context to use for incoming calls fro FWD & Iconnect. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: 12 December 2003 08:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] simple question on sip.conf SW wrote: > Hi folks, > > I want to fix hole in my asterisk set up. > > I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, > Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go > 'other' places. This senario works fine. > > Now the issue is someone else running a vocal or another SIP proxy can > redirect his calls to my * as well. Those calls two will come through > general section of the sip.conf and could land on the PSTN as well. (do not > ask my *'s IP address folks, I am not going to run a free PSTN g/w :) > > So, how do I prevent other than my own proxy to use the general section of > the sip.conf file ? As a mater of fact all calls from fwd and iconnect two > land on the general section. > > Hope some one can shead some light. > > Cheers > > SW > > Here is my sip.conf > > [general] > ;calls arrive from sip lands here > port=5060 > context=default-in Change this to context=forbidden And define "forbidden" context in extensions conf to whatever you like, example exten => s,1,playback(tt-monkeys) exten => s,2,hangup > disallow=all > allow=ulaw > allow=alaw > allow=g729 > maxexpirey=180 > defaultexpirey=160 > ;Connect to Free World Dialup (no NAT) > register=61358:[EMAIL PROTECTED]/61358 This one does not have a defined context. [fwd.pulver.com] section missing > ;Connect to iconnect > register=15108688610:[EMAIL PROTECTED]/15108688610 > canreinvite=no Same here. > > > > [iconnect] > ;incoming does not land here, why ? outgoing is fine > type=friend > secret= > username= > host=sipauth.deltathree.com > dtmfmode=inband ; required by iconnect > context=iconnect-in > canreinvite=no > allow=alaw > allow=ulaw > allow=g729 > > > [fwd] > ;incoming does not land here, why ? outgoing is fine > type=friend > secret=xxx > username=61358 > host=fwd.pulver.com > context=fwd-in > allow=alaw > allow=ulaw > > [vocal] > ;used when dialed in from vocal not working > type=friend > host=ip of vocal server > disallow=all > allow=g729 > allow=ulaw > allow=alaw > port=5060 > canreinvite=no > context=vocal-in > > > [vocal-out] > ;used to dial out to vocal > type=friend > host=ip of vocal server > allow=g729 > allow=ulaw > allow=alaw > port=5065 > canreinvite=no > > [6300] > type=friend > username=6300 > context=intern > ;secret=blah > host=dynamic > ;defaultip=192.168.254.4 > dtmfmode=info > nat=1 > > [6301] > type=friend > username=6301 > host=dynamic > dtmfmode=inband > context=intern > nat=1 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > -- *** Olle E. Johansson, [EMAIL PROTECTED] Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net Runbovägen 10, 192 48 Sollentuna, Sweden Phone: +46 8 594 78 810, Fax: +46 8 594 78 820 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simple question on sip.conf
Hi all Disregard my last post I replied to the wrong e-mail, I should have replied to an off list e-mail. That will teach me not to put my glasses on. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: 12 December 2003 08:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] simple question on sip.conf SW wrote: > Hi folks, > > I want to fix hole in my asterisk set up. > > I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, > Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go > 'other' places. This senario works fine. > > Now the issue is someone else running a vocal or another SIP proxy can > redirect his calls to my * as well. Those calls two will come through > general section of the sip.conf and could land on the PSTN as well. (do not > ask my *'s IP address folks, I am not going to run a free PSTN g/w :) > > So, how do I prevent other than my own proxy to use the general section of > the sip.conf file ? As a mater of fact all calls from fwd and iconnect two > land on the general section. > > Hope some one can shead some light. > > Cheers > > SW > > Here is my sip.conf > > [general] > ;calls arrive from sip lands here > port=5060 > context=default-in Change this to context=forbidden And define "forbidden" context in extensions conf to whatever you like, example exten => s,1,playback(tt-monkeys) exten => s,2,hangup > disallow=all > allow=ulaw > allow=alaw > allow=g729 > maxexpirey=180 > defaultexpirey=160 > ;Connect to Free World Dialup (no NAT) > register=61358:[EMAIL PROTECTED]/61358 This one does not have a defined context. [fwd.pulver.com] section missing > ;Connect to iconnect > register=15108688610:[EMAIL PROTECTED]/15108688610 > canreinvite=no Same here. > > > > [iconnect] > ;incoming does not land here, why ? outgoing is fine > type=friend > secret= > username= > host=sipauth.deltathree.com > dtmfmode=inband ; required by iconnect > context=iconnect-in > canreinvite=no > allow=alaw > allow=ulaw > allow=g729 > > > [fwd] > ;incoming does not land here, why ? outgoing is fine > type=friend > secret=xxx > username=61358 > host=fwd.pulver.com > context=fwd-in > allow=alaw > allow=ulaw > > [vocal] > ;used when dialed in from vocal not working > type=friend > host=ip of vocal server > disallow=all > allow=g729 > allow=ulaw > allow=alaw > port=5060 > canreinvite=no > context=vocal-in > > > [vocal-out] > ;used to dial out to vocal > type=friend > host=ip of vocal server > allow=g729 > allow=ulaw > allow=alaw > port=5065 > canreinvite=no > > [6300] > type=friend > username=6300 > context=intern > ;secret=blah > host=dynamic > ;defaultip=192.168.254.4 > dtmfmode=info > nat=1 > > [6301] > type=friend > username=6301 > host=dynamic > dtmfmode=inband > context=intern > nat=1 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > -- *** Olle E. Johansson, [EMAIL PROTECTED] Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net Runbovägen 10, 192 48 Sollentuna, Sweden Phone: +46 8 594 78 810, Fax: +46 8 594 78 820 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simple question on asterisk
Its all about how you configure your dialplan. Asterisk doesn't know what a PSTN or VOIP phone number is. If you want all 08444 numbers to go through a certain trunk, then you set your dialplan up accordingly. -Kerry > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Mark Hayward > Sent: Monday, March 20, 2006 8:21 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] simple question on asterisk > > Hi, > I am planning to deploy an asterisk installation but I need > to convince a few managers that its a good idea. > Theres something I don't quite understand though, I plan > deploy a box on the end of 4 channel BRI ISDN and provide it > an ADSL internet connection. > Should a phone behind the asterisk PBX wish to call a VOIP > phone number number, say an 0844 one from www.voip-user.org, > would it send this automatically over the PSTN ISDN network > or would it know to send the call over the internet. > Would I need a SIP provider on the internet to forward the > calls? I assume I would need some sort of directory service > to know where to route the call. > Thanks in advance, > Mark > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple question about SIP community
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote: we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. thanks, The generic term for this is 'presence'. Everyone seems to agree that it's important, but I'm not aware of anyone actively working on it for Asterisk. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple question about SIP community
> Something the user list in > Microsoft Messenger. I was thinking on some sort of web page that can > check the registration of the sip clients on the asterisk but want to > know if already exist to avoid to reinvent the wheel. That is actually quite easy and there are some projects that achive this using the Manager API of Asterisk. One is Flash based, but very pretty. I also added rudimentary support for this in DeStar, it has to made nicer and more usable, but that is easy to do. Maybe you visit the page Software Addons on the www.voip-info.org WIKI. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple question about SIP community
Have you had chance to look at Jeff Pulver's Communicator? This is a soft-phone, currently in beta, that allows you to bring together your contacts from MSN, ICQ, AOL and, importantly from your point of view, add contacts that are SIP users. I've not tried it yet with asterisk, but now you have asked the question, I'll try it out... It certainly detects FWD presence so I think it might work with Asterisk. If it doesn't I'll ask put it forward as a suggestion. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: September 09, 2004 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simple question about SIP community On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote: > we have a community of people on an * box that use SIP softphones to > talk each > other. Can you suggest me the quickest and simple way to let someone > know who > is online without have to call one by one the persons to look if they > are > present or not?? Something the user list in Microsoft Messenger. > I was thinking on some sort of web page that can check the > registration of the > sip clients on the asterisk but want to know if already exist to avoid > to > reinvent the wheel. > thanks, The generic term for this is 'presence'. Everyone seems to agree that it's important, but I'm not aware of anyone actively working on it for Asterisk. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple question about SIP community
Marcello Lupo wrote: Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. thanks, Bye, MArcello I would suggest you check out the Flash Operator Panel at www.asternic.org/ . It gives you an overview of who is on the phone and what lines/channels are in use. If you configure it properly, you can even use it to make internal calls. Just simply click on the person you want to talk to, and both of your phones will start ringing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple question about ringing multiple phones (extensions)?
Martin Joseph wrote:
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a
PSTN line from one location to another. I would like to have the
phones at both locations ring when the PSTN # is dialed(inbound calls
from PSTN to asterisk).
I tried something like:
exten => 2020,2,Dial(SIP/2005,25,tr&IAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL=${PHONE_601}&${PHONE_602}&${PHONE_603}
PHONE_601=SIP/601; office 601 Ronald
PHONE_602=SIP/602; office 602 Ronald
PHONE_603=ZAP/1r1; living room 603 cordless
For you this should work too:
exten => 2020,2,Dial(SIP/2005&IAX/2010,25,tr)
bye
Ronald Wiplinger
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RE: [Asterisk-Users] Simple question about ringing multiple phones(extensions)?
Marty, Just remove the options for each technology. Dial(SIP/2005&IAX/2010,25,tr) This should do the job Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: zaterdag 28 januari 2006 9:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Simple question about ringing multiple phones(extensions)? Hey Gurus, I have a very simple asterisk setup that basically lets me share a PSTN line from one location to another. I would like to have the phones at both locations ring when the PSTN # is dialed(inbound calls from PSTN to asterisk). I tried something like: exten => 2020,2,Dial(SIP/2005,25,tr&IAX/2010,25,tr) I thought this might cause both 2005 and 2010 to ring when 2020 was dialed, but only 2005 rings? Thanks for ideas or suggestions on this. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple question about ringing multiple phones (extensions)?
On Jan 28, 2006, at 12:54 AM, Ronald Wiplinger wrote:
Martin Joseph wrote:
I tried something like:
exten => 2020,2,Dial(SIP/2005,25,tr&IAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL=${PHONE_601}&${PHONE_602}&${PHONE_603}
PHONE_601=SIP/601; office 601 Ronald
PHONE_602=SIP/602; office 602 Ronald
PHONE_603=ZAP/1r1; living room 603 cordless
For you this should work too:
exten => 2020,2,Dial(SIP/2005&IAX/2010,25,tr)
Thanks very much for the help guys!
Marty
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