Re: [asterisk-users] SPA112: one analog phone works, not the other
After hours of testing, adding a comma into my FXS port dialplan made it. This obviously relates to some timout handling but beside that, I still have to understand why this seems mandatory. Regards 2014-10-03 11:33 GMT+02:00 Olivier oza.4...@gmail.com: Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the same SPA112 port, I can receive calls (from SIP to analog) but cannot send (from analog to SIP) : nothing shows at asterisk console. When connecting this A400 handset to my provider box (which also has an FXS port), I can successfully send and receive. From this, I conclude my A400 works but differently from my other handset. Basically, when dialing out with my A400, I'm observing this: - I dial my full number (eg 0123456789) then press Send key (as with a mobile phone), - then I hear a long dialing tone from the SPA112 (unplugging the cable between both cut this tone off), - then I hear dialing tones back (those are sent quite fast, one tone for each dialed digit), - then I hear a busy tone and nothing shows at asterisk console. Which SPA112 settings shall I change to get this A400 to work ? What would you suggest ? Regards -- _ -- 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 regcontext (for DUNDi)
On Sun, Oct 5, 2014 at 11:33 AM, Dan Ballance tzewang.do...@gmail.com wrote: Hi, Was this question not appropriate for asterisk-users maybe? Should I post in dev instead? No, I think the question is appropriate for this list. The lack of immediate response probably had more to do with asking it on a Friday than anything else :-) On 4 Oct 2014 15:48, Dan Ballance tzewang.do...@gmail.com wrote: Hi guys, I'm building a PoC Asterisk 12 cluster based on a number of guides I've found on the net. The basic concept is using ARA in conjunction with DUNDi. I have set up ARA with pjsip according to this excellent guide here: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime This is working nicely, so now I am turning my attention to DUNDi, as per this guide here: http://www.ntegratedsolutions.com/wp-content/uploads/2012/07/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf Its seems a really neat solution and I'm keen to implement something similar, however I believe it was written before the pjsip channel driver and I've hit a potential issue I think. The guides for configuring DUNDi seem to suggest using regcontext in sip.conf: [general] regcontext=sipregistration You don't have to use 'regcontext' in order to have the equivalent functionality. regcontext is an automatic mechanism to create a NoOp extension matching each peer's name in a particular context. If you wanted the same functionality for DUNDi, you could create the same extensions for each endpoint in pjsip.conf. However I can't seem to find an equivalent declaration for pjsip.conf. So my questions are: 1) Is there a way to achieve the same functionality with pjsip? Not automatically, no. If you wanted DUNDi to use a context with extensions named the same as the endpoints for DUNDi lookups, you would need to create them yourself. 2) Is DUNDi still being maintained and used? If so, then how should it be configured with modern versions of Asterisk? DUNDi hasn't changed much; hence, the configuration of things for DUNDi is generally the same between recent versions of Asterisk. The DUNDi specific portions of configuration would also be similar between chan_sip and chan_pjsip. As far as 'maintained and used': * DUNDi (which is provided by the pbx_dundi module) is in extended support. That means most development activities for it come from the Asterisk community as a whole. There is not a lot of active development that occurs in pbx_dundi, although patches for bugs are occasionally merged for it. * DUNDi is used by a number of community members, and has some interesting applications in certain setups. I don't have any hard numbers on how many people use it, other than it comes up from time to time in the issue tracker (which is about the extent of my visibility for usage). 3) If DUNDi is not really used in modern set-ups, then what are my alternatives? I really have searched and read and Googled everything I can but I can't seem to find anything on configuring DUNDi with pjsip. Hoping one of you people can point me in the right direction! You may be the first person to try this out! Matt -- 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 channel musicclass from AGI
Hi Matt,
So this actually works (haven't had a chance to try it)?
SET VARIABLE CHANNEL(musicclass) default
Because musicclass is piece of channel information.
Referencing ${musicclass} is not the same thing.
Thanks.
-- James
On Sun, Oct 5, 2014 at 8:05 PM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
Since SetMusicOnHold() is being deprecated, how do we set the channel
musicclass from an AGI script?
Last time I checked you can't call dialplan functions from AGI.
Actually, you can. Any time you can evaluate or set a channel
variable, you can also evaluate or set a dialplan function. Hence, you
can use both 'get variable' [1] or 'set variable' [2]. You could also
use 'exec' and call the Set dialplan application directly.
[1] https://wiki.asterisk.org/wiki/display/AST/AGICommand_get+variable
[2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable
--
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
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Re: [asterisk-users] Pjsip and regcontext (for DUNDi)
On 6 October 2014 15:31, Matthew Jordan mjor...@digium.com wrote: On Sun, Oct 5, 2014 at 11:33 AM, Dan Ballance tzewang.do...@gmail.com wrote: Hi, Was this question not appropriate for asterisk-users maybe? Should I post in dev instead? No, I think the question is appropriate for this list. The lack of immediate response probably had more to do with asking it on a Friday than anything else :-) Thanks ever so much for the informative response Matt. And apologies about the bump over a weekend. I was working over the weekend and forget other people have a better work-life balance than me! Your input is greatly appreciated :) On 4 Oct 2014 15:48, Dan Ballance tzewang.do...@gmail.com wrote: Hi guys, I'm building a PoC Asterisk 12 cluster based on a number of guides I've found on the net. The basic concept is using ARA in conjunction with DUNDi. I have set up ARA with pjsip according to this excellent guide here: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime This is working nicely, so now I am turning my attention to DUNDi, as per this guide here: http://www.ntegratedsolutions.com/wp-content/uploads/2012/07/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf Its seems a really neat solution and I'm keen to implement something similar, however I believe it was written before the pjsip channel driver and I've hit a potential issue I think. The guides for configuring DUNDi seem to suggest using regcontext in sip.conf: [general] regcontext=sipregistration You don't have to use 'regcontext' in order to have the equivalent functionality. regcontext is an automatic mechanism to create a NoOp extension matching each peer's name in a particular context. If you wanted the same functionality for DUNDi, you could create the same extensions for each endpoint in pjsip.conf. I see. I suppose the nice thing about regcontext in combination with the realtime architecture is it means you don't have to hard code this stuff. I guess my next step then is to look at creating these NoOp extensions via the database. However I can't seem to find an equivalent declaration for pjsip.conf. So my questions are: 1) Is there a way to achieve the same functionality with pjsip? Not automatically, no. If you wanted DUNDi to use a context with extensions named the same as the endpoints for DUNDi lookups, you would need to create them yourself. 2) Is DUNDi still being maintained and used? If so, then how should it be configured with modern versions of Asterisk? DUNDi hasn't changed much; hence, the configuration of things for DUNDi is generally the same between recent versions of Asterisk. The DUNDi specific portions of configuration would also be similar between chan_sip and chan_pjsip. As far as 'maintained and used': * DUNDi (which is provided by the pbx_dundi module) is in extended support. That means most development activities for it come from the Asterisk community as a whole. There is not a lot of active development that occurs in pbx_dundi, although patches for bugs are occasionally merged for it. * DUNDi is used by a number of community members, and has some interesting applications in certain setups. I don't have any hard numbers on how many people use it, other than it comes up from time to time in the issue tracker (which is about the extent of my visibility for usage). 3) If DUNDi is not really used in modern set-ups, then what are my alternatives? I really have searched and read and Googled everything I can but I can't seem to find anything on configuring DUNDi with pjsip. Hoping one of you people can point me in the right direction! You may be the first person to try this out! Thanks again for this info Matt. Glad to hear that DUNDi is still maintained. We already have OpenSIPS doing SIP signalling on our network and I'm trying to avoid needing to add more complexity to that set-up. I really love the idea of clustered peer-to-peer asterisk instances sharing registration information with one another - this seems a really neat design to me. If I manage to get this proof of concept running I'll blog about it somewhere so others can learn from my experience. If we actually go ahead and deploy Asterisk using this technique in production then I will make sure we help with the maintenance where we can, all the best, Dan. -- _ -- 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] new app_swift is live
Darren has recently merged my changes to app_swift - now supporting asterisk 12 and 13. If anyone has the Cepstral TTS engine installed and would like to link it with asterisk, app_swift is the way to go. This is the first version that 'configures' to make a Makefile. Please give it a try and report back any issues. git clone 'https://github.com/darrensessions/app_swift' cd app_swift configure make make install make reload -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
