Hi, [wholesale] exten => _1473.,1,Dial(DAHDI/G1/${EXTEN:1}) exten => _1473.,n,Hangup exten => _11473.,1,Dial(DAHDI/G2/${EXTEN:2}) exten => _11473.,n,Hangup Note : Your are sending calls From your MVTS send an extra 1 for 2nd Group. Rgds Mesbah On 5/21/10, Dave George <dgeorge at teletoneinc.com> wrote: > > 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 > > -- > _____________________________________________________________________ > -- 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100521/2b5032b6/attachment-0001.htm