Re: [asterisk-users] Monitoring second leg being dialed?
The image you provided didn't open so I'm not sure about the design. If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll help understand the problem. On Fri, Sep 16, 2011 at 1:28 AM, Gilles codecompl...@free.fr wrote: Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the number I wish to dial, and have it dial out through the FXO card and the FXS port on the ADSL modem. Here's the diagram: http://img844.imageshack.us/img844/3308/asterisksippstncallback.png Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1) when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the call answered although there's no actual phone connection yet, and 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was answered (and if yes, by a live human being rather than an answering machine) or if the line is still ringing. A so-so solution is to simply tell Asterisk to loop through a voice message (This is a call from Joe Allen. Please hit any key and you will be connected), so we know that a human being has answered the call, but I was wondering if there were a better solution. Is it possible for Asterisk to somehow play on channel #1 what's happening on channel #2 while Dahdi/Zaptel is actually still dialing, so that I handle call progression manually from my cellphone and the callee doesn't end up hearing that odd recorded message? 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 -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com wrote: The image you provided didn't open so I'm not sure about the design. Sorry about that. It's a PNG file and it opens in the two browsers I tried. The reason I don't simply get a subscription with a VoIP provider and must go through an Asterisk server + connection to the FXS port is that outgoing calls are free, which is nice when calling cellphones, especially when travelling. If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll help understand the problem. I haven't done it yet, so have no logs to show. I'd simply like to hear what's going on channel #2 while Dahdi is still dialing, instead of simply being kept waiting. Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at home connected to their ADSL modem so that they can make free calls from overseas? 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
Re: [asterisk-users] Monitoring second leg being dialed?
The image just don't open for me, a wild from appears and tells me Domain blocked bla bla. Try attaching image in this mail. Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at home connected to their ADSL modem so that they can make free calls from overseas? LOL- Its like asking an army Have you guys ever worked with guns!! :P Please try producing SIP traces so your problem could be identified. Which asterisk and DAHDI version you are using btw? On Fri, Sep 16, 2011 at 2:51 PM, Gilles codecompl...@free.fr wrote: On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com wrote: The image you provided didn't open so I'm not sure about the design. Sorry about that. It's a PNG file and it opens in the two browsers I tried. The reason I don't simply get a subscription with a VoIP provider and must go through an Asterisk server + connection to the FXS port is that outgoing calls are free, which is nice when calling cellphones, especially when travelling. If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll help understand the problem. I haven't done it yet, so have no logs to show. I'd simply like to hear what's going on channel #2 while Dahdi is still dialing, instead of simply being kept waiting. 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 -- _ -- 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] Monitoring second leg being dialed?
Hi Gilles, Sorry about that. It's a PNG file and it opens in the two browsers I tried. It opens here too. It's very simple though. I would put it like this: VOIP phone ---SIP over the internet--- Asterisk ---internal FXO card--- PSTN-outlet ---PSTN--- PSTN phone Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at home connected to their ADSL modem so that they can make free calls from overseas? I have asimilar situation. Except, I don't have an internal card but an external SPA3102 -box which converts my PSTN line to SIP and connects that to my Asterisk box. The SPA3102 was far cheaper than any FXO card I could find. I have other SIP (hard or soft) phones connect to my Asterisk as well. Some on the LAN, some over internet. I can use all the phones to dial out on the PSTN line and incoming calls just the same (in real life I only allow a few phones to call out and only one or two will ring when a call come in). I think this is a very common situation, so I'm not really sure what your problem is. Perhaps it's because I don't use an internal card, but in my situation it works just fine. I dial a number on my SIP phone, Asterisk goes through the dialplan, and puts the call out via the SPA3102. In my ear I hear ringing sounds, busy, wrong number or someone talking to me just like if I had connected a normal phone to the PSTN line. I'm probably not understanding why you think that the FXO card should provide you with the call progress status. I don't think it works like that on a PSTN line, you just have to listen to the sounds yourself. Best regards, Jeroen Eeuwes -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: I think this is a very common situation, so I'm not really sure what your problem is. Perhaps it's because I don't use an internal card, but in my situation it works just fine. I dial a number on my SIP phone, Asterisk goes through the dialplan, and puts the call out via the SPA3102. In my ear I hear ringing sounds, busy, wrong number or someone talking to me just like if I had connected a normal phone to the PSTN line. I haven't done this yet, and was looking for information. I was under the (apparently false) impression that Asterisk/Dahdi didn't connect the two legs until the callee had gone off-hook. So it looks like there's really no issue in connecting a remote SIP client with a PSTN number through an FXO port + ADSL VoIP modem to take advantage of free phone calls. Thanks everyone. -- _ -- 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] Monitoring second leg being dialed?
On 09/16/2011 06:13 AM, Gilles wrote: On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: I think this is a very common situation, so I'm not really sure what your problem is. Perhaps it's because I don't use an internal card, but in my situation it works just fine. I dial a number on my SIP phone, Asterisk goes through the dialplan, and puts the call out via the SPA3102. In my ear I hear ringing sounds, busy, wrong number or someone talking to me just like if I had connected a normal phone to the PSTN line. I haven't done this yet, and was looking for information. I was under the (apparently false) impression that Asterisk/Dahdi didn't connect the two legs until the callee had gone off-hook. This is true, but you already answered your own question in your original post: since Asterisk cannot know whether the called party (dialing out via an FXO port) has answered or not, it assumes the outgoing call is 'answered' as soon as dialing has been completed. Because of this, the calling channel is bridged to the called channel as soon as dialing has been completed, and the calling party will hear the progress of the outbound call. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming kpflem...@digium.com wrote: This is true, but you already answered your own question in your original post: since Asterisk cannot know whether the called party (dialing out via an FXO port) has answered or not, it assumes the outgoing call is 'answered' as soon as dialing has been completed. Because of this, the calling channel is bridged to the called channel as soon as dialing has been completed, and the calling party will hear the progress of the outbound call. Thanks for the confirmation. Too bad Dahdi doesn't provide call supervision so that Asterisk knows if/when the callee has answered. I'll experiment and see how it goes. -- _ -- 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] Monitoring second leg being dialed?
It does on PRI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Friday, September 16, 2011 7:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Monitoring second leg being dialed? On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming kpflem...@digium.com wrote: This is true, but you already answered your own question in your original post: since Asterisk cannot know whether the called party (dialing out via an FXO port) has answered or not, it assumes the outgoing call is 'answered' as soon as dialing has been completed. Because of this, the calling channel is bridged to the called channel as soon as dialing has been completed, and the calling party will hear the progress of the outbound call. Thanks for the confirmation. Too bad Dahdi doesn't provide call supervision so that Asterisk knows if/when the callee has answered. I'll experiment and see how it goes. -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 19:35:19 -0400, Eric Wieling ewiel...@nyigc.com wrote: It does on PRI. Unfortunately, this is for an ADSL modem, hence the connection to its FXS port :-/ -- _ -- 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] Monitoring second leg being dialed?
Thanks for the confirmation. Too bad Dahdi doesn't provide call supervision so that Asterisk knows if/when the callee has answered. I'll experiment and see how it goes. DAHDI with an FXO card can support call answer/hangup supervison. Check out chan_dahdi.conf options; answeronpolarityswitch=yes hanguponpolarityswitch=yes However for this to work with an FXO card, the line from the telco does need to provide the line reversal. OT but related, an FXS port can also provide answer/hangup line reversal for an upstream device, by setting the same options above for an FXS port. Alec -- _ -- 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] Monitoring second leg being dialed?
Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the number I wish to dial, and have it dial out through the FXO card and the FXS port on the ADSL modem. Here's the diagram: http://img844.imageshack.us/img844/3308/asterisksippstncallback.png Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1) when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the call answered although there's no actual phone connection yet, and 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was answered (and if yes, by a live human being rather than an answering machine) or if the line is still ringing. A so-so solution is to simply tell Asterisk to loop through a voice message (This is a call from Joe Allen. Please hit any key and you will be connected), so we know that a human being has answered the call, but I was wondering if there were a better solution. Is it possible for Asterisk to somehow play on channel #1 what's happening on channel #2 while Dahdi/Zaptel is actually still dialing, so that I handle call progression manually from my cellphone and the callee doesn't end up hearing that odd recorded message? 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
