Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP
Hello Mathew, Thank you for the reply. I will open an issue and send debug information. Can you explain more about the workaround? A reference to the documentation would be fine. Thanks again, Yaron. On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan mjor...@digium.com wrote: On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? No one that I'm aware of is currently working on that. As Asterisk is an open source project, if having the 'auto' feature added to the PJSIP stack is something you're interested in, you should consider writing a patch for the project [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk does not transcode the DTMF signals, therefore DTMF is not working. It used to work on release 11. This is really bad. Do you know of a solution to this issue? Maybe some settings? That actually is a bug. You are most likely ending up in a native packet to packet bridge (or a native remote bridge), which does not decode the RTP stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is being passed to the other side. Please do open an issue for that [2]. Make sure you provide a full DEBUG log, as that will illustrate what is actually occurring. Note that you can work around that issue by adding a feature flag to whatever application caused the bridging to occur. [2] https://issues.asterisk.org/jira -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- _ -- 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] Questions on musiconhold.conf custom mode
2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com: Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Is the gsm-format an option for you? So you may convert your moh-File to gsm: sox YouWavFile.wav -r 8000 -c1 MohFile.gsm Hi Thorsten, Yes gsm-format is an option for me but how can you play such gsm file as MOH ? If I'm not mistaken, both madplay or mpg123 would only play MP3 files (I've not tested with other formats, yet). I could successfully play a RAW file with cat but cat has no repeat option, so I still have to find something else anyway. If you really need mp3 you have to compile sox with mp3-support by yourself OR maybe this is a solution on Debian: http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/ Yes, you're correct. I'll suggest my customer a Wheezy upgrade. -Thorsten- -- _ -- 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
[asterisk-users] Detect hangup due to RTP timeout
Hello, Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached? Thank you, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Questions on musiconhold.conf custom mode
Am 27.10.2014 08:54, schrieb Olivier: 2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com: Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Is the gsm-format an option for you? So you may convert your moh-File to gsm: sox YouWavFile.wav -r 8000 -c1 MohFile.gsm Hi Thorsten, Yes gsm-format is an option for me but how can you play such gsm file as MOH ? If I'm not mistaken, both madplay or mpg123 would only play MP3 files (I've not tested with other formats, yet). I could successfully play a RAW file with cat but cat has no repeat option, so I still have to find something else anyway. When your musiconhold.conf looks like that ... cut - [general] [default] mode=files directory=moh [your_moh_class] mode=files directory=/your/path/to/your/moh/files cut - ... then you can put any supported file format into the specified directory. GSM is only one option. Asterisk will take the best (meaning cheapest) file format availble in this directory. If you really need mp3 you have to compile sox with mp3-support by yourself OR maybe this is a solution on Debian: http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/ Yes, you're correct. I'll suggest my customer a Wheezy upgrade. -- _ -- 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] make asterisk do something when an outgoing call is picked up
Am 26.10.2014 00:43, schrieb lee:
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called that the call will be
recorded.
Here's what I'm trying to do:
call comes in:
if(I pick up) {
play announcement to caller;
start recording;
let me talk to the caller;
end recording when call ends;
send recording to my email account;
} else {
record voice mail;
}
call goes out:
if(call is picked up) {
play announcement to callee;
if(callee hangs up) {
end call;
} else {
start recording;
let me talk to callee;
end recording when call ends;
send recording to my email account;
}
} else {
call ends;
offer me to automatically call again later;
}
Please keep in mind that I'm new to asterisk and just got it to work.
Searching for having asterisk do something when an outgoing call is
picked up has been unsuccessful other than that I found out that you can
have it make outgoing calls automatically to play pre-recorded messages:
So asterisk does have a way to detect when a call is picked up and a way
of doing something when that happens.
What I have working so far is incoming and outgoing calls and voicemail
for one phone/user, which is a basic set up I'm trying extend and
improve now.
Maybe this will do a good job for recording all calls:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
And playing an announcement, when a call is picked, should be done
within your dialplan with this function:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Playback
-Thorsten-
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[asterisk-users] authentication time for asterisk server
Hi all, what should i do if i want to know how long asterisk server take a time for registration 1 client on server side? especially just for voip server authentication, when we have to registered username and password in sip.conf and extensions.conf files? it's not how long registration for one client authenticate to voip server. i really need to know, and really thankful for any helps. kind regards alf -- _ -- 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] DTMF behavior in asterisk 12 with PJSIP
On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello Mathew, Thank you for the reply. I will open an issue and send debug information. Can you explain more about the workaround? A reference to the documentation would be fine. Sure - really, what you are running into is a difference in how Asterisk bridges channels: https://wiki.asterisk.org/wiki/display/AST/Bridges I suspect the reason DTMF is not decoded is because you are in a native bridge (local or remote). You can force a core two-party bridge by requiring that Asterisk decode the media and detect DTMF. Those requirements are done by setting the various 'feature' flags on whatever dialplan application is causing the channels to be bridged. For an example, see the 't' or 'T' options in Dial: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] Setting Music on Hold with the Manager Interface
On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote: Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this. # Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); Use the CHANNEL function: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Action: Setvar Channel: (your channel name here) Variable: CHANNEL(musicclass) Value: (your MoH class here) -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] Setting Music on Hold with the Manager Interface
Thanks Matt. I tried that already, no luck. Still, I get blank nothingness instead of MOH. I will try again just to be sure I didn't miss something. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Date: Mon, 27 Oct 2014 08:51:42 -0500 Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface From: mjor...@digium.com To: tjrl...@live.com; asterisk-users@lists.digium.com On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote: Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); Use the CHANNEL function: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Action: SetvarChannel: (your channel name here)Variable: CHANNEL(musicclass)Value: (your MoH class here) -- Matthew Jordan Digium, Inc. | Engineering Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USACheck us out at: http://digium.com http://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
Re: [asterisk-users] Asterisk 12 Dialplan
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Friday, October 24, 2014 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan
On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota mgandik...@nts.net
wrote:
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects there
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel
from Alice enters into extension 1000. This prints out all the
information about her channels. After hearing silence for a while, she
hangs up - and our script notifies us that her channel has left the
application.
end of quote
Is there some way the call can be moved to the next priority or context
in the dial plan from the stasis app? It seems the caller is stuck in
stasis.
Once a channel hangs up it is controlled by hangup handlers and h
extens.
If however you want to kick an active channel out of your stasis
application
to run dialplan then you use the
POST /channels/{channelId}/continue
ARI command.
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API
#Asterisk12ChannelsRESTAPI-continueInDialplan
Thanks, Richard. How do I get manager events such as VarSetEvent
(https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_Var
Set) using ARI?
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Re: [asterisk-users] Asterisk 12 Dialplan
On Mon, Oct 27, 2014 at 10:56 AM, Murthy Gandikota mgandik...@nts.net wrote: -- Thanks, Richard. How do I get manager events such as VarSetEvent ( https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_VarSet) using ARI? Events are provided by your WebSocket connection - a good overview of how this works is provided on the wiki [1]. You will receive events for resources that you are subscribed to; you are automatically subscribed to any channel that enters your Stasis application [2]. You can create subscriptions to things outside of your application using the applications resource [3]. The possible events are all documented in the data models [4]. Specifically, however, an AMI VarSet event corresponds to an ARI ChannelVarSet event [5]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Stasis [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Applications+REST+API [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models#Asterisk13RESTDataModels-ChannelVarset -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] AppKonference 2.6
I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonference That said Asterisk 13 doesn’t get that much attention because I use Asterisk 1.4 + some hacks. Here’s a link to my Asterisk 1.4 github repository: https://github.com/pjalbrecht/asterisk You don’t need these hacks to use the module, but you may find them useful.-- _ -- 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] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)
The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that it would be “better for a small select group of users at the expense of the majority of community that use the product as designed for the purpose it was originally intended. However, you’re either very naive or delusional if you think the community is going to follow you down that path. Do you really believe the community is going simply chuck their dial plans and walk away from their investment in Asterisk? Not likely, dude. On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie j...@ocjtech.us wrote: On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht palbre...@glccom.com wrote: When Matt says deprecating the dial plan would be difficult and would take a long time it seems to me he’s being evasive and misleading. He doesn’t say it’s never going to happen and he doesn’t share whatever he thinks the Asterisk vision actually is which he should presumably be aware of since he is the Asterisk engineering manager. Why do you keep insisting that Digium promise to *never* deprecate dial plans? I don't think that's a promise that's really worth anything as there may be really good reasons in the future to do so. I think that you've gotten the best that you will get: they've said that there are no plans within Digium to deprecate the dial plan, and if there were plans, they'd give people a long time prepare before it actually happens. It's probably a good time to refresh your understanding of Digium's support policies: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Version 13 will be around until at least 2018, so you'll have *at least* that long to prepare for the switch, since version 13 is feature frozen so there's no way the dial plan would be removed from 13. And all of this talk of deprecating the dial plan isn't even coming from Digium. It's something that was suggested by a community member at the developer conference. I wasn't there so I don't know how seriously it was taken there, but it would have been impolite of everyone involved to just ignore it. -- Jeff Ollie -- _ -- 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] [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)
On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote: The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that it would be “better for a small select group of users at the expense of the majority of community that use the product as designed for the purpose it was originally intended. However, you’re either very naive or delusional if you think the community is going to follow you down that path. Do you really believe the community is going simply chuck their dial plans and walk away from their investment in Asterisk? Not likely, dude. My comment/question wasn't really about dial plans, per se. My question was about you insisting that Digium make such unqualified promises about the future of Asterisk. Even though Digium is a private company, I believe that they are still bound by U.S. laws regarding forward-looking statements[1]. So even if they wanted to (which I doubt), there's no way you're going to get the promise that you're looking for. [1] http://en.wikipedia.org/wiki/Forward-looking_statement -- Jeff Ollie -- _ -- 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] Asterisk 12 Dialplan
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Monday, October 27, 2014 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12 Dialplan On Mon, Oct 27, 2014 at 10:56 AM, Murthy Gandikota mgandik...@nts.net wrote: Thanks, Richard. How do I get manager events such as VarSetEvent (https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_Var Set) using ARI? Events are provided by your WebSocket connection - a good overview of how this works is provided on the wiki [1]. You will receive events for resources that you are subscribed to; you are automatically subscribed to any channel that enters your Stasis application [2]. You can create subscriptions to things outside of your application using the applications resource [3]. The possible events are all documented in the data models [4]. Specifically, however, an AMI VarSet event corresponds to an ARI ChannelVarSet event [5]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Stasi s [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Applications+REST +API [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models# Asterisk13RESTDataModels-ChannelVarset -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org I am unable to detect the Manager_Setvar event using ARI. Can you please let me know, in ARI lingo, the curl or javascript code to detect the AMI Manager_Setvar event for myvar in the following dialplan: [default] exten = 1000,1,NoOp() same = n,Answer() same = n,set(myvar=test) same = n,Stasis(hello-world) same = n,Hangup() Thanks -- _ -- 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] Asterisk 12 Dialplan
On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota mgandik...@nts.net wrote: I am unable to detect the Manager_Setvar event using ARI. Can you please let me know, in ARI lingo, the curl or javascript code to detect the AMI Manager_Setvar event for myvar in the following dialplan: [default] exten = 1000,1,NoOp() same = n,Answer() same = n,set(myvar=test) same = n,Stasis(hello-world) same = n,Hangup() Thanks Perhaps it would be easier if you provided some information about the ARI application you've written. Have you connected a WebSocket? Are you receiving other ARI events? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] sip.conf to pjsip.conf conversion script
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip
I assume I run it from /etc/asterisk with the input and output file as
arguments however there's no instructions and I don't Grok python.
Unfortunately it's not working, Despite what the below error states I do
have a udpbindaddr set to 0.0.0.0 in my configuration.
root@kiniston01:/etc/asterisk#
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
sip.conf pjsip.conf
Traceback (most recent call last):
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1158, in module
pjsip, non_mappings = convert(sip, pjsip_filename, dict(), False)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1090, in convert
map_transports(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 817, in map_transports
create_udp(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 590, in create_udp
bind = sip.multi_get('general', ['udpbindaddr', 'bindaddr'])[0]
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/astconfigparser.py,
line 407, in multi_get
(key_list, section))
LookupError: keys ['udpbindaddr', 'bindaddr'] not found for section
'general'
I've not turned up anything useful with Google so the mailing list is my
next step.
I can provide my configuration if needed however it is just the stock
sip.conf with a phone and two trunks added at the bottom.
Thanks!
--
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
--
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Re: [asterisk-users] Asterisk 12 Dialplan
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Monday, October 27, 2014 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12 Dialplan On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota mgandik...@nts.net wrote: I am unable to detect the Manager_Setvar event using ARI. Can you please let me know, in ARI lingo, the curl or javascript code to detect the AMI Manager_Setvar event for myvar in the following dialplan: [default] exten = 1000,1,NoOp() same = n,Answer() same = n,set(myvar=test) same = n,Stasis(hello-world) same = n,Hangup() Thanks Perhaps it would be easier if you provided some information about the ARI application you've written. Have you connected a WebSocket? Are you receiving other ARI events? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org I am using ari4java to capture stasis events like StasisStart, StatisEnd, etc. However, I am unable to capture the Varset event as explained before. In particular the myvar variable is not associated with any app It is perhaps a channel variable. -- _ -- 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
