[asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH

[Danny Nicholas]

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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday, September 26, 2012 10:15 AM
To: asterisk-users at 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-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul 
> Belanger
> Sent: Wednesday, September 26, 2012 7:52 AM
> To: asterisk-users at 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 at voipphones:1] Set("SIP/4856-00000003",
>> "CALLERID(num)=xxxxxxxxx") in new stack
>>       -- Executing [97052660 at 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.belanger at 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.belanger at 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