Re: [asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]
Nevermind,
I checked the code, and A* is not using the F option in MeetMe for
Page(), so it's not working by default.
Attached is a patch which fixes the problem for me, if anyone needs it.
Matteo
Il 11/02/2012 13:53, Matteo Fortini ha scritto:
Noone knows that? Where/whom could I ask?
Thanks
Il 10/02/2012 12:30, Matteo Fortini ha scritto:
Hi,
I'd like to implement some way of controlling remote SIP clients
while in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is
connected to the previous one
I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the
caller to the callees. I found option 'F' for MeetMe, but I have no
control on Page().
TIA,
Matteo
diff -Nurd asterisk-1.6.2.20.orig/apps/app_page.c asterisk-1.6.2.20/apps/app_page.c
--- asterisk-1.6.2.20.orig/apps/app_page.c 2009-01-25 14:35:48.0 +0100
+++ asterisk-1.6.2.20/apps/app_page.c 2012-02-13 13:47:53.509396266 +0100
@@ -164,7 +164,7 @@
timeout = atoi(args.timeout);
}
- snprintf(meetmeopts, sizeof(meetmeopts), MeetMe,%ud,%s%sqxdw(5), confid, (ast_test_flag(flags, PAGE_DUPLEX) ? : m),
+ snprintf(meetmeopts, sizeof(meetmeopts), MeetMe,%ud,%s%sqxFdw(5), confid, (ast_test_flag(flags, PAGE_DUPLEX) ? : m),
(ast_test_flag(flags, PAGE_RECORD) ? r : ) );
/* Count number of extensions in list by number of ampersands + 1 */
@@ -247,7 +247,7 @@
}
if (!res) {
- snprintf(meetmeopts, sizeof(meetmeopts), %ud,A%s%sqxd, confid, (ast_test_flag(flags, PAGE_DUPLEX) ? : t),
+ snprintf(meetmeopts, sizeof(meetmeopts), %ud,A%s%sqFxd, confid, (ast_test_flag(flags, PAGE_DUPLEX) ? : t),
(ast_test_flag(flags, PAGE_RECORD) ? r : ) );
pbx_exec(chan, app, meetmeopts);
}
--
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Re: [asterisk-users] DTMF forwarding and Page
Noone knows that? Where/whom could I ask? Thanks Il 10/02/2012 12:30, Matteo Fortini ha scritto: Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the callees. I found option 'F' for MeetMe, but I have no control on Page(). TIA, Matteo -- _ -- 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] DTMF forwarding and Page
Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the callees. I found option 'F' for MeetMe, but I have no control on Page(). TIA, Matteo -- _ -- 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 Playback sound dropping on linphone
There are some other clients, even if they are mainly testing/demo applications for some SIP stacks. sofsip-cli for SofiaSIP (which is backed by Nokia) simpleopal for OpalVOIP They do work, even if they're not as full featured as linphone in some ways, e.g. on soundcard management. They offer some more options in other fields. Regarding the Playback issue, it seems that Playback into a [ConfBridge|MeetMe] conference stutters and drops randomly. I think I'll file a bug for that. Thank you Il 12/11/2010 10:23, Sebastian ha scritto: Hi On 11/11/2010 03:35 PM, Matteo Fortini wrote: Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... I use linphonec as well - and haven't found another console sip phone either. I'd be interested if there is another one. Sebastian Thank you, Matteo -- _ -- 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] Asterisk Playback sound dropping on linphone
Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo -- _ -- 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 Playback sound dropping on linphone
I did some more tests, and it's not really a problem with linphone: the rtp capture shows empty packets sent by Asterisk. Since the channel which is doing Playback() is in a MeetMe conference, I tried also to speak on another phone on the same conference: well the rtp capture shows the stream from A* becoming silent, then the new sound from the phone comes up. Do I have to file a bug? Thank you, Matteo Il 11/11/2010 16:35, Matteo Fortini ha scritto: Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo -- _ -- 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] Asterisk linphone call dropping by itself
hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two linphones register with a name which is defined as dynamic in sip.conf the call terminates on the caller's side, while the callee is still connected, and I have to force the termination on that side. I'm using asterisk 1.8.0 and linphone 3.99 I really don't know how to investigate further... a capture on sip ports just shows that on the 25th ack packet the other side answers with a BYE instead of with an OK SDP packet. TIA, Matteo -- _ -- 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] [SOLVED][BUG??] Asterisk linphone call dropping by itself
Well the problem seems to be: the linphones are listening on port 5062, while * is on port 5060. For some reason, the INVITEs are received from *, but are forwarded on port 5060 by default. I solved the problem by moving * to port 5062 and moving the linphones back to port 5060. All is well, but may this be a bug? Thanks, M Il 03/11/2010 12:48, Matteo Fortini ha scritto: hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two linphones register with a name which is defined as dynamic in sip.conf the call terminates on the caller's side, while the callee is still connected, and I have to force the termination on that side. I'm using asterisk 1.8.0 and linphone 3.99 I really don't know how to investigate further... a capture on sip ports just shows that on the 25th ack packet the other side answers with a BYE instead of with an OK SDP packet. TIA, Matteo -- _ -- 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] Page minimum number of extensions
Hi, if I Page more than one extension, then the MeetMe conference stays up even if all the called extensions aren't available or are hung up. Is there a way of keeping track of how many extensions are attached to the conference, and require a number or a particular extension to be present? Thank you -- _ -- 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] Same extension on multiple servers confusion
Hi, I have the same extension registered with multiple softphones on multiple servers, i.e. 100-lo...@hosta 100-lo...@hostb and on both hostA and hostB I have the extension in extension.conf exten = 100,1,Answer() exten = 100,n,Dial(100-local) When from softphone registered as 100-lo...@hosta I call (1...@hostb) what I see on the softphone on host B is 100-lo...@hostb is contacting you and the call gets routed on the local calls context instead of the incoming call context. I expected to see 100-lo...@hosta is contacting you instead. Is this behavior something that can be avoided? I thought it would be normal to have two asterisk's in e.g. two companies serving the same extensions... Thank you, Matteo -- _ -- 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 as a distributed paging system
Hi! thank you for your good answers. Another related question: I tried using Page() and it works perfectly, but I need to implement a slightly different behavior, and I'm looking into ways of implementing it. When a user picks up the phone and chooses to page the speakers, the call should start (so that it's ready for talking), but in muted status. When the user pushes a push-to-talk button, then a bell sound needs to be played through all the speakers, then she can start talking freely. Everytime the PTT button is released, the mic needs to mute, but that's something I can work out in the softphone. How can I implement it? I am thinking of using some call parking method and some DTMF code to pass to the next state, but I am open to advice, since I'm quite new to Asterisk. Could I also create a macro to do the same thing Page is doing, but with ConfBridge? Last question: is there a way of reinviting periodically remotes to the conference, so that they can recover after e.g. a reboot? Thank you in advance, Matteo Il 22/09/2010 21:51, Philipp von Klitzing ha scritto: Hi! I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. Most of the major phone vendors (that are employed by the users of this list) have support for multi-cast of some sort. In recent firmware release notes I have read that SNOM has now also added a feature to feed multicast directly from a phone (and not just play multicast audio on the speaker as long as the phone is not in use). I'm currently using some software I wrote which sends voice over multicast RTP and coordinates all the sites with multicast messages. app_page has been around for quite some in Asterisk, and the new Asterisk 1.8 now also adds the channel driver MulticastRTP. Is there a way asterisk could be of use, or would I need to bend it too much? Look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom I have made good experience with MAST for multicasting SNOM phones: http://www.aelius.com/njh/mast/ Philipp -- _ -- 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] Asterisk as a distributed paging system
I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm currently using some software I wrote which sends voice over multicast RTP and coordinates all the sites with multicast messages. I don't own the switches so each site will be assigned an address by DHCP, that's why I'm using multicast. I heard of asterisk and SIP as a possible alternative to my software, and I'd rather use tested and widely adopted software. Is there a way asterisk could be of use, or would I need to bend it too much? Thank you in advance, Matteo -- _ -- 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
