Re: [Asterisk-Users] Zapata FXO always answers call?
Reid A. Forrest wrote: Ummm X100P IS an analog adaptor. Well, I thought he was talking about some of the ATA´s available on the market. If an X100P is one analog adapter and has this problem, then everything else has this problem too (except E1/T1/PRI boards that have it´s own signaling). Ok, so how you guys set up voicemail or anything else that needs more than one priority level? I can´t believe nobody has voicemail on analog lines!! That would be a major flaw in asterisk. As I and others have said, callprogress is experimental and of not much use. For me it does nothing. Gelson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Monday, June 07, 2004 4:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zapata FXO always answers call? Steven Critchfield wrote: your using an analog adapter. We cover this regularly. You need callprogress detection since you don't have a reliable way of doing it via the analog adapter. No, I´m not using an analog adapter. As I said, my X100P connects directly to my PBX. Gelson On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote: I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it´s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it´s still ringing. The problem is that now I want to set up voicemail to those analog extensions, but since * says it answered on first ring it never goes to the next priority, where voicemail is called. I tried callprogress=yes on zapata.conf but it has no effect. Here is a tipical log from a call I have _not_ answered: -- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack -- Called g1/32 -- Zap/1-1 answered SIP/2000-a638 I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a workaround for it? Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zapata FXO always answers call?
Steven Critchfield wrote: your using an analog adapter. We cover this regularly. You need callprogress detection since you don't have a reliable way of doing it via the analog adapter. No, I´m not using an analog adapter. As I said, my X100P connects directly to my PBX. Gelson On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote: I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it´s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it´s still ringing. The problem is that now I want to set up voicemail to those analog extensions, but since * says it answered on first ring it never goes to the next priority, where voicemail is called. I tried callprogress=yes on zapata.conf but it has no effect. Here is a tipical log from a call I have _not_ answered: -- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack -- Called g1/32 -- Zap/1-1 answered SIP/2000-a638 I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a workaround for it? Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zapata FXO always answers call?
Ummm X100P IS an analog adaptor. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Monday, June 07, 2004 4:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zapata FXO always answers call? Steven Critchfield wrote: your using an analog adapter. We cover this regularly. You need callprogress detection since you don't have a reliable way of doing it via the analog adapter. No, I´m not using an analog adapter. As I said, my X100P connects directly to my PBX. Gelson On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote: I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it´s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it´s still ringing. The problem is that now I want to set up voicemail to those analog extensions, but since * says it answered on first ring it never goes to the next priority, where voicemail is called. I tried callprogress=yes on zapata.conf but it has no effect. Here is a tipical log from a call I have _not_ answered: -- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack -- Called g1/32 -- Zap/1-1 answered SIP/2000-a638 I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a workaround for it? Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata FXO always answers call?
I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it´s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it´s still ringing. The problem is that now I want to set up voicemail to those analog extensions, but since * says it answered on first ring it never goes to the next priority, where voicemail is called. I tried callprogress=yes on zapata.conf but it has no effect. Here is a tipical log from a call I have _not_ answered: -- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack -- Called g1/32 -- Zap/1-1 answered SIP/2000-a638 I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a workaround for it? Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zapata FXO always answers call?
your using an analog adapter. We cover this regularly. You need callprogress detection since you don't have a reliable way of doing it via the analog adapter. On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote: I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it´s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it´s still ringing. The problem is that now I want to set up voicemail to those analog extensions, but since * says it answered on first ring it never goes to the next priority, where voicemail is called. I tried callprogress=yes on zapata.conf but it has no effect. Here is a tipical log from a call I have _not_ answered: -- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack -- Called g1/32 -- Zap/1-1 answered SIP/2000-a638 I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a workaround for it? Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users