[asterisk-ss7] app_dial.c:1518 dial_exec_full: Unable to create channel of type 'DAHDI'

Dave George dgeorge at teletoneinc.com
Fri May 21 05:11:08 CDT 2010


Thanks for the suggestions.  If I have two groups setup, how can I split
the calls between the two groups?  Is there a dial(DAHDI) option to
choose the groups randomly?

Dave George
Teletone Inc.


> -------- Original Message --------
> Subject: Re: [asterisk-ss7] app_dial.c:1518 dial_exec_full: Unable to
> create channel of type 'DAHDI'
> From: Mesbahuddin Malik <mesbah.malik at gmail.com>
> Date: Fri, May 21, 2010 6:00 am
> To: asterisk-ss7 at lists.digium.com
> 
> 
>    Hi,
> 
>    Can  you make a try  with a  different group for
> 
>    group=2
>    slc=1
>    sigchan = 73
>    cicbeginswith = 126
>    channel = 74-96
> 
>    Rgds
>    Mesbah
> 
>    On 5/21/10, Dave George <dgeorge at teletoneinc.com> wrote:
>      Hi Malik,
>      When the first T1 is full, calls to the second T1 fails.  Second T1
>      full, calls to first fails.  Off peak hours I can make a call on any T1.
>      See the logs below
>      Is there some varial in chan_dahdi that could be limiting me to 1 T1.
>      In the logs I don't see any SS7 call setup messages so I doubt this is
>      coming from the other end.
>         -- Hungup 'DAHDI/24-1'
>      == Spawn extension (wholesale, 14734436295, 1) exited non-zero on
>      'SIP/MVTS2-00aa1e18'
>      == Using SIP RTP CoS mark 5
>      == Using UDPTL CoS mark 5
>         -- Executing [14734380035 at wholesale:1] Dial("SIP/MVTS-a18060c8",
>      "DAHDI/g1/4734380035") in new stack
>         -- Called g1/4734380035
>         -- DAHDI/24-1 is proceeding passing it to SIP/MVTS-a18060c8
>         -- DAHDI/24-1 is ringing
>         -- DAHDI/24-1 answered SIP/MVTS-a18060c8
>      == Using SIP RTP CoS mark 5
>      == Using UDPTL CoS mark 5
>         -- Executing [14734352124 at wholesale:1] Dial("SIP/MVTS-a178fd48",
>      "DAHDI/g1/4734352124") in new stack
>      [May 20 19:39:47] WARNING[12578]: app_dial.c:1518 dial_exec_full: Unable
>      to create channel of type 'DAHDI' (cause 34 - Circuit/channel
>      congestion)
>      == Everyone is busy/congested at this time (1:0/1/0)
>         -- Executing [14734352124 at wholesale:2] Hangup("SIP/MVTS-a178fd48",
>      "") in new stack
>      == Spawn extension (wholesale, 14734352124, 2) exited non-zero on
>      'SIP/MVTS-a178fd48'
>      == Using SIP RTP CoS mark 5
>      == Using UDPTL CoS mark 5
>         -- Executing [14734352124 at wholesale:1] Dial("SIP/MVTS2-a22ff068",
>      "DAHDI/g1/4734352124") in new stack
>      [May 20 19:39:47] WARNING[12579]: app_dial.c:1518 dial_exec_full: Unable
>      to create channel of type 'DAHDI' (cause 34 - Circuit/channel
>      congestion)
>      == Everyone is busy/congested at this time (1:0/1/0)
>         -- Executing [14734352124 at wholesale:2] Hangup("SIP/MVTS2-a22ff068",
>      "") in new stack
>      == Spawn extension (wholesale, 14734352124, 2) exited non-zero on
>      'SIP/MVTS2-a22ff068'
>      == Using SIP RTP CoS mark 5
>      == Using UDPTL CoS mark 5
>         -- Executing [14734352124 at wholesale:1] Dial("SIP/MVTS2-a155e628",
>      "DAHDI/g1/4734352124") in new stack
>      [May 20 19:39:48] WARNING[12580]: app_dial.c:1518 dial_exec_full: Unable
>      to create channel of type 'DAHDI' (cause 34 - Circuit/channel
>      congestion)
>      == Everyone is busy/congested at this time (1:0/1/0)
>         -- Executing [14734352124 at wholesale:2] Hangup("SIP/MVTS2-a155e628",
>      "") in new stack
>      == Spawn extension (wholesale, 14734352124, 2) exited non-zero on
>      'SIP/MVTS2-a155e628'
>      == Using SIP RTP CoS mark 5
>      == Using UDPTL CoS mark 5
>         -- Executing [14734352124 at wholesale:1] Dial("SIP/MVTS-a170fd08",
>      "DAHDI/g1/4734352124") in new stack
>      [May 20 19:39:48] WARNING[12581]: app_dial.c:1518 dial_exec_full: Unable
>      to create channel of type 'DAHDI' (cause 34 - Circuit/channel
>      congestion)
>      == Everyone is busy/congested at this time (1:0/1/0)
>         -- Executing [14734352124 at wholesale:2] Hangup("SIP/MVTS-a170fd08",
>      "") in new stack
>      == Spawn extension (wholesale, 14734352124, 2) exited non-zero on
>      'SIP/MVTS-a170fd08'
>                             SCCP method indicator: 0
>                             [ 54 06 ]
>         -- DAHDI/13-1 is proceeding passing it to SIP/MVTS-a23f0f38
>         -- DAHDI/13-1 is ringing
>      Len = 19 [ 97 f5 10 a5 01 9d 02 00 a3 01 15 75 00 0c 02 00 02 80 90 ]
>      FSN: 117 FIB 1
>      BSN: 23 BIB 1
>      <[0] MSU
>      [ 97 f5 10 ]
>             Network Indicator: 2 Priority: 2 User Part: ISUP (5)
>             [ a5 ]
>             OPC 1-163-0 DPC 2-157-1 SLS 21
>             [ 01 9d 02 00 a3 01 15 ]
>                     CIC: 117
>                     [ 75 00 ]
>                     Message Type: REL
>                     [ 0c ]
>                     --VARIABLE LENGTH PARMS[1]--
>                     Cause Indicator:
>                             Coding Standard: 0
>                             Location: 0
>                             Cause Class: 1
>                             Cause Subclass: 0
>                             Cause: Normal call clearing (16)
>                             [ 02 80 90 ]
>      Len = 14 [ f5 98 0b b5 00 a3 01 01 9d 02 c2 75 00 10 ]
>      FSN: 24 FIB 1
>      BSN: 117 BIB 1
>      >[0] MSU
>      [ f5 98 0b ]
>             Network Indicator: 2 Priority: 3 User Part: ISUP (5)
>             [ b5 ]
>             OPC 2-157-1 DPC 1-163-0 SLS 194
>             [ 00 a3 01 01 9d 02 c2 ]
>                     CIC: 117
>                     [ 75 00 ]
>                     Message Type: RLC
>                     [ 10 ]
>         -- Hungup 'DAHDI/17-1'
>      Dave George
>      Teletone Inc.
>      > -------- Original Message --------
>      > Subject: Re: [asterisk-ss7] app_dial.c:1518 dial_exec_full: Unable to
>      > create channel of type 'DAHDI'
>      > From: Mesbahuddin Malik <mesbah.malik at gmail.com>
>      > Date: Fri, May 21, 2010 5:22 am
>      > To: asterisk-ss7 at lists.digium.com
>      >
>      >
>      > ---------------------------------------------------------------------
>      > --
>      > _____________________________________________________________________
>      > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>      >
>      > asterisk-ss7 mailing list
>      > To UNSUBSCRIBE or update options visit:
>      >    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>      --
>      _____________________________________________________________________
>      -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>      asterisk-ss7 mailing list
>      To UNSUBSCRIBE or update options visit:
>        http://lists.digium.com/mailman/listinfo/asterisk-ss7
> ---------------------------------------------------------------------
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-ss7


More information about the asterisk-ss7 mailing list