[asterisk-users] PJSIP and Multiple transports per endpoint
I have a multihomed machine. How can I assign multiple IPs to and endpoint, not all of them, just two, for instance, out of many? Suppose the machine as 30 IPs, but my asterisk needs listen on two, and one single endpoint needs to be associated with those two IPs. I tried to add a second bind line to a transport, but it ignores all after the first one. I tried to add a second transport line to an endpoint, but it only considers one. Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Multiple transports per endpoint
CDR wrote: I have a multihomed machine. How can I assign multiple IPs to and endpoint, not all of them, just two, for instance, out of many? Suppose the machine as 30 IPs, but my asterisk needs listen on two, and one single endpoint needs to be associated with those two IPs. I tried to add a second bind line to a transport, but it ignores all after the first one. I tried to add a second transport line to an endpoint, but it only considers one. Thanks for your help. The transport line controls what transport is used for outgoing traffic to an endpoint. It is used when the code chooses the wrong one using automatic logic. What do you mean by assign multiple IPs to an endpoint. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel h323 and oh323 fails to match inbound IP
I am having the issue described in this question: http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html Does anybody has an insight? I guess Asterisk is trying to match the combination IP:Port, but in H223 this changes call by call. There is no way to add insecure=port like in channel_sip. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP and Multiple transports per endpoint
I had some confusion here. The endpoint needs a transport in order to carry calls out. But the transports are also used by the application PJSIP at large, in order to listen for incoming connections. In order to just receive calls, I think you only need a transport, but no need to assign that transport to any endpoint. For example if you are just acting as voicemail or a pure IVR system. If you have a multi-homed machine, you need a transport for each IP where you expect to receive calls. Please correct me if I am wrong. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pattern Extension not working in Dialplan
Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten = s,1,Background(my/age) ;;Play recorded message to enter age exten = s,n,WaitExten(10) exten = _XX,1,Set(AGE=${EXTEN});; this line is not executing, instead dialplan is terminating with error given below. exten = s,n,NoOp(${AGE}) exten = s,n,GotoIf($[${LEN(${AGE})} 0]?notEmpty) exten = s,n,Goto(s,1) exten = s(notEmpty),n,Background(my/thank-you) exten = s,n,Wait(1) When I receive call and tries to enter the digits (86 lets say), it only accept just first digit and terminates even before considering second digit. Error message : WARNING[5726][C-000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please suggest what might be wrong. Anurag Rana http://newbie42.blogspot.in/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern Extension not working in Dialplan
The first issue I see is you are attempting to insert your pattern match in the middle of your 's' extension, That's going to break your 's' extension. The second issue is that you are matching on XX which will match two digits, You need to match on _X instead if you are attempting to match on the number 8. I recommend you look into 'read' instead of trying to do a pattern match. On Sun, Sep 7, 2014 at 1:41 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten = s,1,Background(my/age) ;;Play recorded message to enter age exten = s,n,WaitExten(10) exten = _XX,1,Set(AGE=${EXTEN});; this line is not executing, instead dialplan is terminating with error given below. exten = s,n,NoOp(${AGE}) exten = s,n,GotoIf($[${LEN(${AGE})} 0]?notEmpty) exten = s,n,Goto(s,1) exten = s(notEmpty),n,Background(my/thank-you) exten = s,n,Wait(1) When I receive call and tries to enter the digits (86 lets say), it only accept just first digit and terminates even before considering second digit. Error message : WARNING[5726][C-000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please suggest what might be wrong. Anurag Rana http://newbie42.blogspot.in/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SIP warning
Hi, upto asterisk 1.8 you used to get this error if there were more than 1 m= line in an invite... Asterisk was just telling you it was declining the second. I belive from 10.0 onwards asterisk now just replies back with port 0 to the stream it isn't interested in... You can ignore it - if its bothering you upgrade to asterisk 11 which is very solid now. On 6 September 2014 10:28, CDR vene...@gmail.com wrote: I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream: audio 30660 RTP/AVP 4 101 13 What does it mean and does it show a problem like one-way audio? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern Extension not working in Dialplan
Please don't top-post. On Sun, Sep 7, 2014 at 1:41 PM, Anurag Rana anuragrana31...@gmail.com wrote: I created a dummy dialplan where I ask the user to enter the age. [macro-age] exten = s,1,Background(my/age) ;;Play recorded message to enter age exten = s,n,WaitExten(10) exten = _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead dialplan is terminating with error given below. exten = s,n,NoOp(${AGE}) exten = s,n,GotoIf($[${LEN(${AGE})} 0]?notEmpty) exten = s,n,Goto(s,1) exten = s(notEmpty),n,Background(my/thank-you) exten = s,n,Wait(1) On Sun, 7 Sep 2014, John Kiniston wrote: The first issue I see is you are attempting to insert your pattern match in the middle of your 's' extension, That's going to break your 's' extension. The second issue is that you are matching on XX which will match two digits, You need to match on _X instead if you are attempting to match on the number 8. I recommend you look into 'read' instead of trying to do a pattern match. A pattern match is a reasonable method. I use pattern matching more often that the read() application. Try both and see which meets your needs better. Are you really defining a 'macro' or is that just the (misleading) name you chose for your context. Personally, I use gosub() more, but again, try both :) I suggest you try 'dialplan show macro-age' to see how Asterisk is interpreting your dialplan. I suspect it is not what you expect. In specific, your ordering of '_xx' in the middle of 's' is odd. This would disrupt the value of the priority in older versions of Asterisk, but it appears that it does work in modern (I'm using 11) versions. Also, a label ('notEmpty') belongs to a priority, not an extension. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern Extension not working in Dialplan
On Sun, 7 Sep 2014, Steve Edwards wrote: In specific, your ordering of '_xx' in the middle of 's' is odd. This would disrupt the value of the priority in older versions of Asterisk, but it appears that it does work in modern (I'm using 11) versions. Disregard that. I can't even follow my own advice ('dialplan show macro-age'). Don't 'intermingle' extensions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfer Fails - Not a Valid Extension
We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message that is not a valid extension. Does anyone have any ideas about where to begin looking for the source of that error? Phil Ledon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern Extension not working in Dialplan
Thank you all for your suggestions. 1. [macro-age] is a macro and not an extension badly named. 2. I am able to use Read to fulfill the purpose but we can't use Read() after Background(). To use read we need Playback() [ am I right?]. But Playback do not provide barge-in facility i.e. user have to listen whole message then only his inputs will be accepted and if he entered input during the time recording is played , the input will be lost. So if using Background() [which return the control immediately] I have to use _XX extension. 3. So basically I want to create a dial-plan where user is asked to input multi-digit value and he can enter it without listening complete message (if the user knows the message already) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users