Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile
Howard Lowndes wrote: On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote: I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension rings for about 5 secs (as long as it takes the TDM400 to dial the mobile number), then just the telstra mobile rings.. From asterisk -vvvr -- Goto (custom-incoming,s,1) -- Executing Dial(SIP/202-b424, Zap/g0/0408xxSip/200|30|t) in new stack -- Called g0/0408xx -- Called 200 -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- Zap/2-1 answered SIP/202-b424 This tend to indicate to me that the mobile system has picked up the call request on the zap channel and that * therefore thinks that the zap channel has picked up the call and will then bridge the zap channel to the sip 202 channel and kill off the ringing on the sip 200 channel. I don't know that there is much you can do about this as basically you are trying to get interaction on two different systems. No. Analog ports are always considered ANSWERED as soon as Asterisk finishes dialing. This is covered over and over and over again in the mailing list archives. There are a few very ugly hacks to work around the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile
Shane Dalgleish wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, 18 February 2005 2:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile Howard Lowndes wrote: On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote: I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension rings for about 5 secs (as long as it takes the TDM400 to dial the mobile number), then just the telstra mobile rings.. From asterisk -vvvr -- Goto (custom-incoming,s,1) -- Executing Dial(SIP/202-b424, Zap/g0/0408xxSip/200|30|t) in new stack -- Called g0/0408xx -- Called 200 -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- Zap/2-1 answered SIP/202-b424 This tend to indicate to me that the mobile system has picked up the call request on the zap channel and that * therefore thinks that the zap channel has picked up the call and will then bridge the zap channel to the sip 202 channel and kill off the ringing on the sip 200 channel. I don't know that there is much you can do about this as basically you are trying to get interaction on two different systems. No. Analog ports are always considered ANSWERED as soon as Asterisk finishes dialing. This is covered over and over and over again in the mailing list archives. There are a few very ugly hacks to work around the problem. Thanks Howard and Eric, I did have a look around for this before I posted and I found a few references to: callprogress=yes (in zapata.conf) But also read that this only (kinda) works in the US. Also had a brief look at BackgroundDetect, but it looks a bit rough What I do need to do urgently however is get rid of the 5 or so seconds of silence and static noise between the time Zap says the call is answered and Telstra establishes the call and starts ringing again.. So what I'm thinking is perhaps:- - Call the phones in the office - Call the mobiles seperately but at the same time - wait for a DMTF tone from the mobile (I think I could put up with that) - bridges the call to the mobile - But if a Sip phone answers the call first hangup on the mobile - bridges the call to the Sip phone Any thoughts on how I would go about that? You can replace the t option with tr at the end of your Dial line. However, if the destination is busy then you may hear a couple of rings and then a busy sound. Why are you using t in the first place? You really do need a PRI or VoIP service provider. These things are not really issues with PRI or VoIP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile
On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote: I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension rings for about 5 secs (as long as it takes the TDM400 to dial the mobile number), then just the telstra mobile rings.. From asterisk -vvvr -- Goto (custom-incoming,s,1) -- Executing Dial(SIP/202-b424, Zap/g0/0408xxSip/200|30|t) in new stack -- Called g0/0408xx -- Called 200 -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- Zap/2-1 answered SIP/202-b424 This tend to indicate to me that the mobile system has picked up the call request on the zap channel and that * therefore thinks that the zap channel has picked up the call and will then bridge the zap channel to the sip 202 channel and kill off the ringing on the sip 200 channel. I don't know that there is much you can do about this as basically you are trying to get interaction on two different systems. At this stage the mobile is still ringing and has not been answered. Below are zapata.conf extensions-custom.conf Any thoughts anyone? Cheers Shane ---zapata.conf--- language=en context=from-pstn signalling=fxs_ks ;stripmsd=1 immediate=no overlapdial=yes faxdetect=no usecallerid=no echocancel=yes callprogress=yes busydetect=yes busycount=6 echocancelwhenbridged=no echotraining=800 rxgain=5.5 group=0 channel=2 channel=3 group=1 usecallerid=yes channel=4 ---extensions-custom.conf--- [custom-incoming] exten = s,1,Dial(Zap/g0/0408xxSip/200,30,t) exten = s,104,Voicemail(u200) exten = s,105,Hangup() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
