[asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! -- _ -- 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] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 7:52 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? /* I'm dialing 8 digits because in my extensions.conf required user to dial 9 for outgoing calls. */ Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. /* thanks for pointing that out, I overlook Cause No. 27. I will check aging my Dahdi configuration */ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
You are set up as a USA PRI, but not dialing a USA TN. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz Sent: Wednesday, September 26, 2012 11:13 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 7:52 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? /* I'm dialing 8 digits because in my extensions.conf required user to dial 9 for outgoing calls. */ Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. /* thanks for pointing that out, I overlook Cause No. 27. I will check aging my Dahdi configuration */ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- _ -- 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] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
On 12-09-26 11:12 AM, motty.cruz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 7:52 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? /* I'm dialing 8 digits because in my extensions.conf required user to dial 9 for outgoing calls. */ Right, but does your CO require you to pass the '9' to them or are you to strip it? Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. /* thanks for pointing that out, I overlook Cause No. 27. I will check aging my Dahdi configuration */ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
You need to modify your dialplan to change 9xxx to 1aaaxxx. I think most U.S. SIP providers want a 10 digit number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 10:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 11:12 AM, motty.cruz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 7:52 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? /* I'm dialing 8 digits because in my extensions.conf required user to dial 9 for outgoing calls. */ Right, but does your CO require you to pass the '9' to them or are you to strip it? Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. /* thanks for pointing that out, I overlook Cause No. 27. I will check aging my Dahdi configuration */ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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
