Re: [Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing - workaround
Hi folks, This is what I am doing at this time : exten => _,1,TrySystem(..command that sends a jabber message..) exten => _,2,Set(calling=${EXTEN:0:4}) exten => _,3,ChanIsAvail(SIP/[EMAIL PROTECTED]) exten => _,4,Dial(SIP/[EMAIL PROTECTED],15,tr) exten => _,5,Goto(_-${DIALSTATUS},1) exten => _,104,Dial(Zap/g2/${calling},15,tr) exten => _,105,Goto(_-${DIALSTATUS},1) exten => _-NOANSWER,1,Voicemail(u${calling}) exten => _-NOANSWER,2,Hangup exten => _-CHANUNAVAIL,1,Voicemail(u${calling}) exten => _-CHANUNAVAIL,2,Hangup exten => _-CONGESTION,1,Voicemail(u${calling}) exten => _-CONGESTION,2,Hangup exten => _-BUSY,1,Voicemail(b${calling}) exten => _-BUSY,2,Hangup exten => _-CANCEL,1,Voicemail(u${calling}) exten => _-CANCEL,2,Hangup Long story short, Asterisk lets me know if the SIP users has their SIP phone on. If it is, I call it. Otherwise I call their POTS. Probably should consider calling the POTS if SIP does not answer in timeout time, but that's for another day :). Thanks everyone for helping out along the way, JES C F wrote: You can try one more thing, and that is the M option, and create a macro that announces to the user to accept the call. as documented at: http://www.voip-info.org/wiki-asterisk+cmd+dial On 11/23/05, James MacLean <[EMAIL PROTECTED]> wrote: Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like "Dial(ZAP/g2/&SIP/[EMAIL PROTECTED])" and have it wait 'til the correct phone is answered :(. Thanks to C F for the "c" option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote: Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. Dial(Zap/1/&SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/&SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
You can try one more thing, and that is the M option, and create a macro that announces to the user to accept the call. as documented at: http://www.voip-info.org/wiki-asterisk+cmd+dial On 11/23/05, James MacLean <[EMAIL PROTECTED]> wrote: > Oh boy :(. > > As Roman politely explained in a private email... I was using ports 1 > and 2 thinking they were the outbound fxs ports :(. That's it, these > glasses are going, and no more testing from home :). When I switched to > testing with ports 3 and 4, everything worked the same as G2. > > Not of course as cute as what I had hoped for when I see the local telco > can do something like "Dial(ZAP/g2/&SIP/[EMAIL PROTECTED])" and have it > wait > 'til the correct phone is answered :(. Thanks to C F for the "c" option > but my goal was to just have the 4 digit number call folks with and > without SIP. I would not expect users to know to press #. I don't think > dvlinedetect will quite cut it either. callprogress looked promising, > but, alas, as many others have found, it hangs up after timeout seconds. > I'll keep digging :). > > Thanks again everyone, > JES > > James B. MacLean wrote: > > > Hi C F, > > > > I am not well versed in this level of telephony or Asterisk, so please > > bare with me :). > > > > My setup is really typical. Bought the digium card with 4 ports. 2 fxs > > / 2 fxo. The 2 fxo's are connected directly to phones, belong to group > > 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. > > > > The 2 fxs ports are connected to the telco, belong to group 2 > > according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. > > > > Dial(Zap/1/&SIP/[EMAIL PROTECTED],15,r) works as expected, > > Dial(Zap/2/&SIP/[EMAIL PROTECTED],15,r) works as expected > > > > but: > > > > Dial(Zap/g2/&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered > > to Asterisk. > > > > Does this support what you are explaining? I'm honestly confused by > > how an fxs module operates as an fxo module? > > > > Thanks for any more direction you might have, > > JES > > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like "Dial(ZAP/g2/&SIP/[EMAIL PROTECTED])" and have it wait 'til the correct phone is answered :(. Thanks to C F for the "c" option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote: Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. Dial(Zap/1/&SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/&SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. Dial(Zap/1/&SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/&SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES C F wrote: I'm going to guess here that the problem is *not* related to using the g. It just happnes to be that on your asterisk system when you use g2 it rings to an FXO module while when using channel 2 on the zap interface you are calling an FXS channel. Please correct me if I'm wrong. Zap FXO modules will consider themselfs answered as soon as it goes of hook (as this is realy called answered, since all the signalling is done inband), as will FXS modules, but FXOs are meant to call ppl using in band signalling, and that is what tells you (the caller) if the person answers or not, but because it uses in band signalling it is actualy answred as soon as you pick up the phone (just like your ananlog handset that is plugged into an FXS module *answers* the line as soon as it goes off hook, even though it is just trying to make a phone call). The only way to work this around is if you use c in the dial command for the zap channel, but then the called person has to press # to connect the call. Hope this helps. On 11/22/05, James B. MacLean <[EMAIL PROTECTED]> wrote: Hi Folks, Took some effort to even realize what was happening and I did not find anything obvious in the archives or FAQs to explain it. Simply put, if I have a dial plan that dials to a specific ZAP line (ie exten => _,n,Dial(Zap/2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) then both the SIP and the ZAP are called and ZAP keeps reporting "ringing" until they are answered or a timeout occurs. This is the correct expectation as I understand it. But... When I change the ZAP channel to use a group (ie exten => _,n,Dial(Zap/g2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) then I always get something like : -- SIP/8438-37ca is ringing -- Zap/3-1 answered SIP/8438-be27 Even though the Zap channel is _not_ answered. The phone does keep ringing and I can complete the call if I pick up the phone. So using single channels (either of the 2 I use) works, but together in a group gives me these results. I am using the latest CVS of everything. Any explanation, workaround greatly appreciated :), JES smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
I'm going to guess here that the problem is *not* related to using the g. It just happnes to be that on your asterisk system when you use g2 it rings to an FXO module while when using channel 2 on the zap interface you are calling an FXS channel. Please correct me if I'm wrong. Zap FXO modules will consider themselfs answered as soon as it goes of hook (as this is realy called answered, since all the signalling is done inband), as will FXS modules, but FXOs are meant to call ppl using in band signalling, and that is what tells you (the caller) if the person answers or not, but because it uses in band signalling it is actualy answred as soon as you pick up the phone (just like your ananlog handset that is plugged into an FXS module *answers* the line as soon as it goes off hook, even though it is just trying to make a phone call). The only way to work this around is if you use c in the dial command for the zap channel, but then the called person has to press # to connect the call. Hope this helps. On 11/22/05, James B. MacLean <[EMAIL PROTECTED]> wrote: > Hi Folks, > > Took some effort to even realize what was happening and I did not find > anything obvious in the archives or FAQs to explain it. > > Simply put, if I have a dial plan that dials to a specific ZAP line (ie > exten => _,n,Dial(Zap/2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) > then both the SIP and the ZAP are called and ZAP keeps reporting > "ringing" until they are answered or a timeout occurs. This is the > correct expectation as I understand it. > > But... When I change the ZAP channel to use a group (ie exten => > _,n,Dial(Zap/g2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) then I > always get something like : > > -- SIP/8438-37ca is ringing > -- Zap/3-1 answered SIP/8438-be27 > > Even though the Zap channel is _not_ answered. The phone does keep > ringing and I can complete the call if I pick up the phone. > > So using single channels (either of the 2 I use) works, but together in > a group gives me these results. I am using the latest CVS of everything. > > Any explanation, workaround greatly appreciated :), > JES > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
Hi Folks, Took some effort to even realize what was happening and I did not find anything obvious in the archives or FAQs to explain it. Simply put, if I have a dial plan that dials to a specific ZAP line (ie exten => _,n,Dial(Zap/2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) then both the SIP and the ZAP are called and ZAP keeps reporting "ringing" until they are answered or a timeout occurs. This is the correct expectation as I understand it. But... When I change the ZAP channel to use a group (ie exten => _,n,Dial(Zap/g2/${calling}&SIP/[EMAIL PROTECTED],15,tr)) then I always get something like : -- SIP/8438-37ca is ringing -- Zap/3-1 answered SIP/8438-be27 Even though the Zap channel is _not_ answered. The phone does keep ringing and I can complete the call if I pick up the phone. So using single channels (either of the 2 I use) works, but together in a group gives me these results. I am using the latest CVS of everything. Any explanation, workaround greatly appreciated :), JES smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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