[asterisk-users] Transer calls hitting #
Hi, Any idears how to get call transfer to work? The "#" key is recognized but the following typed digits does not appear to be read and the IVR announce "Invald extension..." Debug output -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1 -- Unable to find extension '' in context 'local_extensions' ... extensions.conf ... [local_extensions] include => outgoing ; Local extensions exten => 1001,1,Dial(SIP/1001,20,rtT) exten => 1002,1,Dial(SIP/1002,20,rtT) exten => 1003,1,Dial(SIP/1003,20,rtT) ... ___ --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] Transer calls hitting #
Poul Moller wrote: Hi, -- Unable to find extension '' in context 'local_extensions' It looks like your phone isn't sending the digits you keyed after you did the transfer. Try turning up your logging. set verbose 20 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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] Transer calls hitting #
You are right kind of. I tried from an IP SIP phone and it worked. The other phones (analog) are all connected via a Linksys pap2 ATA adapter. All (and different) alalog phones behave similar. The # key is recognized but the others aren't. Both the IPhone and ATA's use G711a codec. Are there any special ATA audio setting I should apply? Poul On 4/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: Poul Moller wrote: > Hi, > > > -- Unable to find extension '' in context 'local_extensions' It looks like your phone isn't sending the digits you keyed after you did the transfer. Try turning up your logging. set verbose 20 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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 ___ --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] Transer calls hitting #
Poul Moller wrote: Are there any special ATA audio setting I should apply? That I don't know, I've never setup an ATA before. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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] Transer calls hitting #
Try to configure your PAP2 DTMF send mode to INFO. On 4/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: Poul Moller wrote: > Are there any special ATA audio setting I should apply? > That I don't know, I've never setup an ATA before. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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 ___ --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] Transer calls hitting #
Getting better... however still l didn't managed to transfer a call from my ATA. As you see some digits new gets recognized but never the full extension (1002 in my case). The # however always correctly triggers the transfer IVR. /Poul SIP/1003-08e68090 is ringing -- SIP/1003-08e68090 answered SIP/1001-08e62b50 -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '02' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '002' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') On 4/22/07, Leonardo Kamache (Gmail) <[EMAIL PROTECTED]> wrote: Try to configure your PAP2 DTMF send mode to INFO. On 4/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: > Poul Moller wrote: > > Are there any special ATA audio setting I should apply? > > > > That I don't know, I've never setup an ATA before. > > Doug > > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." > > > ___ > --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 > ___ --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 ___ --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] Transer calls hitting #
Hello there, you should check how the ATA encodes DTMF tones (eg rfc2833), and that you have the same setting in sip.conf. l. In data Sat, 21 Apr 2007 18:21:34 +0200, Poul Moller <[EMAIL PROTECTED]> ha scritto: You are right kind of. I tried from an IP SIP phone and it worked. The other phones (analog) are all connected via a Linksys pap2 ATA adapter. All (and different) alalog phones behave similar. The # key is recognized but the others aren't. Both the IPhone and ATA's use G711a codec. Are there any special ATA audio setting I should apply? Poul On 4/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: Poul Moller wrote: > Hi, > > > -- Unable to find extension '' in context 'local_extensions' It looks like your phone isn't sending the digits you keyed after you did the transfer. Try turning up your logging. set verbose 20 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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 -- Home of QueueMetrics - http://queuemetrics.com ___ --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] Transer calls hitting #
Maybe you have too short a digit timeout. l. In data Sun, 22 Apr 2007 11:39:59 +0200, Poul Moller <[EMAIL PROTECTED]> ha scritto: Getting better... however still l didn't managed to transfer a call from my ATA. As you see some digits new gets recognized but never the full extension (1002 in my case). The # however always correctly triggers the transfer IVR. /Poul SIP/1003-08e68090 is ringing -- SIP/1003-08e68090 answered SIP/1001-08e62b50 -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '02' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/1003-08e68090' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/1003-08e68090 -- Unable to find extension '002' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') On 4/22/07, Leonardo Kamache (Gmail) <[EMAIL PROTECTED]> wrote: Try to configure your PAP2 DTMF send mode to INFO. On 4/21/07, Doug Lytle <[EMAIL PROTECTED]> wrote: > Poul Moller wrote: > > Are there any special ATA audio setting I should apply? > > > > That I don't know, I've never setup an ATA before. > > Doug > > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." > > > ___ > --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 > ___ --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 -- Home of QueueMetrics - http://queuemetrics.com ___ --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
