You need to modify your dialplan to change 9xxxxxxx to 1aaaxxxxxxx. 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/4856-00000003", >> "CALLERID(num)=xxxxxxxxx") in new stack >> -- Executing [97052660@voipphones:2] Dial("SIP/4856-00000003", >> "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/4856-00000003' 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