I tend to start getting those errors after a call has been made via that channel
Kind regards On Fri, Dec 17, 2010 at 7:44 AM, Edrich de Lange <[email protected]> wrote: > Both connect to the same platform (erricson) > > Also, On my side it says the links are up. but the remote side not. > > ss7.conf > > [linkset-mtnR1] > ; The linkset is enabled > enabled => yes > > ; The end-of-pulsing (ST) is not used to determine when incoming > address is complete > enable_st => no > > ; Reply incoming call with CON rather than ACM and ANM > use_connect => no > > ; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even > CIC numbers, most recently used > hunting_policy => seq_lth > > ; Incoming calls are placed in the ss7 context in the asterisk dialplan > context => mtn > > ; The language for this context is da > language => da > > ; The value and action for t35. Value is in msec, action is either st or > timeout > ; If you use overlapped dialling dial plan, you might choose: t35 => 4000,st > t35 => 15000,timeout > > ; The subservice field: national (8), international (0), auto or > decimal/hex value > ; The auto means that the subservice is obtained from first received SLTM > subservice => auto > > ; The host running the mtp3 service > ; mtp3server => localhost > ; SS7 variant, either ITU or CHINA > variant => ITU > > ; The point code for this SS7 signalling point is 0x8e0 > ; If point code is included here, it must not occur in host section > opc => 720 > > ; The destination point (peer) code is 0x3fff for linkset mtnR1 > ; If point code is included here, it must not occur in host section > dpc => 1392 > > [linkset-mtnJ1] > ; The linkset is enabled > enabled => yes > > ; The end-of-pulsing (ST) is not used to determine when incoming > address is complete > enable_st => no > > ; Reply incoming call with CON rather than ACM and ANM > use_connect => no > > ; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even > CIC numbers, most recently used > hunting_policy => seq_lth > > ; Incoming calls are placed in the ss7 context in the asterisk dialplan > context => mtn > > ; The language for this context is da > language => da > > ; The value and action for t35. Value is in msec, action is either st or > timeout > ; If you use overlapped dialling dial plan, you might choose: t35 => 4000,st > t35 => 15000,timeout > > ; The subservice field: national (8), international (0), auto or > decimal/hex value > ; The auto means that the subservice is obtained from first received SLTM > subservice => auto > > > ; The host running the mtp3 service > ; mtp3server => localhost > ; SS7 variant, either ITU or CHINA > variant => ITU > > ; The point code for this SS7 signalling point is 0x8e0 > ; If point code is included here, it must not occur in host section > opc => 720 > > ; The destination point (peer) code is 0x3fff for linkset mtnR1 > ; If point code is included here, it must not occur in host section > dpc => 1368 > > > > [link-l1] > > ; This link belongs to linkset mtnR1 > linkset => mtnJ1 > > ; The speech/audio circuit channels on this link > channels => 1-15,17-31 > > ; The signalling channel > schannel => 16 > ; To use the remote mtp3 service, use 'schannel => remote,16' > > ; The first CIC > firstcic => 33 > > ; The link is enabled > enabled => yes > ; Echo cancellation > ; echocancel can be one of: no, 31speech (enable only when > transmission medium is 3.1Khz speech), allways > echocancel => no > ; echocan_train specifies training period, between 10 to 100 msec > echocan_train => 350 > ; echocan_taps specifies number of taps, 32, 64, 128 or 256 > echocan_taps => 128 > ; RX and TX gains > rxgain => 0.0 > > txgain => 0.0 > ; Relax DTMF, yes or no > relaxdtmf => no > ; If link is connected to an STP with point code 0x3ff0, the following > may be needed > stp => 1832 > > > [link-l2] > > ; This link belongs to linkset mtnR1 > linkset => mtnR1 > > ; The speech/audio circuit channels on this link > channels => 1-15,17-31 > > ; The signalling channel > schannel => 16 > ; To use the remote mtp3 service, use 'schannel => remote,16' > > ; The first CIC > firstcic => 1 > > ; The link is enabled > enabled => yes > > ; Echo cancellation > ; echocancel can be one of: no, 31speech (enable only when > transmission medium is 3.1Khz speech), allways > echocancel => no > ; echocan_train specifies training period, between 10 to 100 msec > echocan_train => 350 > ; echocan_taps specifies number of taps, 32, 64, 128 or 256 > echocan_taps => 128 > ; RX and TX gains > rxgain => 0.0 > > txgain => 0.0 > ; Relax DTMF, yes or no > relaxdtmf => no > ; If link is connected to an STP with point code 0x3ff0, the following > may be needed > stp => 2832 > > [host-xtrj1] > ; chan_ss7 auto-configures by matching the machines host name with the > host-<name> > ; section in the configuration file, in this case 'gentoo1'. The same > ; configuration file can thus be used on several hosts. > > ; The host is enabled > enabled => yes > > > > ; Syntax: links => link-name:digium-connector-no > ; The links on the host is 'l1', connected to span/connector #1 > links => l1:1,l2:2 > > ; The SCCP global title: translation-type, nature-of-address, > numbering-plan, address > globaltitle => 0x00, 0x04, 0x01, 4546931411 > ssn => 7 > > > [jitter] > ;------------------------------ JITTER BUFFER CONFIGURATION > -------------------------- > jbenable = no ; Enables the use of a jitterbuffer on the > receiving side of a > ; SIP channel. Defaults to "no". An > enabled jitterbuffer will > ; be used only if the sending side can > create and the receiving > ; side can not accept jitter. The SIP > channel can accept jitter, > ; thus a jitterbuffer on the receive SIP > side will be used only > ; if it is forced and enabled. > > ; jbforce = no ; Forces the use of a jitterbuffer on > the receive side of a SIP > ; channel. Defaults to "no". > > ; jbmaxsize = 200 ; Max length of the jitterbuffer in > milliseconds. > > ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over > which the jitterbuffer is > ; resynchronized. Useful to improve the > quality of the voice, with > ; big jumps in/broken timestamps, > usually sent from exotic devices > ; and programs. Defaults to 1000. > > ; jbimpl = fixed ; Jitterbuffer implementation, used on > the receiving side of a SIP > ; channel. Two implementations are > currently available - "fixed" > ; (with size always equals to jbmaxsize) > and "adaptive" (with > ; variable size, actually the new jb of > IAX2). Defaults to fixed. > > ; jblog = no ; Enables jitterbuffer frame logging. > Defaults to "no". > ;----------------------------------------------------------------------------------- > > > And the linestat > > Linkset: mtnR1 > CIC 1 Idle > CIC 2 Idle > CIC 3 Idle > CIC 4 Idle > CIC 5 Idle > CIC 6 Idle > CIC 7 Idle > CIC 8 Idle > CIC 9 Idle > CIC 10 Idle > CIC 11 Idle > CIC 12 Idle > CIC 13 Idle > CIC 14 Idle > CIC 15 Idle > CIC 17 Idle > CIC 18 Idle > CIC 19 Idle > CIC 20 Idle > CIC 21 Idle > CIC 22 Idle > CIC 23 Idle > CIC 24 Idle > CIC 25 Idle > CIC 26 Idle > CIC 27 Idle > CIC 28 Idle > CIC 29 Idle > CIC 30 Idle > CIC 31 Idle > Linkset: mtnJ1 > CIC 33 Busy > CIC 34 Idle > CIC 35 Idle > CIC 36 Idle > CIC 37 Idle > CIC 38 Idle > CIC 39 Idle > CIC 40 Idle > CIC 41 Idle > CIC 42 Idle > CIC 43 Idle > CIC 44 Idle > CIC 45 Idle > CIC 46 Idle > CIC 47 Idle > CIC 49 Idle > CIC 50 Idle > CIC 51 Idle > CIC 52 Idle > CIC 53 Idle > CIC 54 Idle > CIC 55 Idle > CIC 56 Idle > CIC 57 Idle > CIC 58 Idle > CIC 59 Idle > CIC 60 Idle > CIC 61 Idle > CIC 62 Idle > CIC 63 Idle > > Kind regards > > Edd > > > On Thu, Dec 16, 2010 at 2:31 PM, Amish Chana <[email protected]> wrote: >> Hi, >> >> Are both your links on the same platform? >> Can you post the output of ss7 linestat and ss7.conf. >> >> A. >> >> >> On 12/15/2010 03:58 PM, Edrich de Lange wrote: >>> >>> Basically what I see as the channels >>> 1-7 up >>> 8-15 Down >>> 16 MTP >>> 17-24 up >>> 25-31 down >>> >>> >>> This seems very simmetrical. >>> >>> Has anyone seen an issue like this? >>> >>> I can send calls via it and all >>> >>> Kind regards >>> >>> On Wed, Dec 15, 2010 at 1:03 PM, Edrich de Lange<[email protected]> wrote: >>>> >>>> all are Idle >>>> >>>> and >>>> >>>> >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=63 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=60 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=62 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=56 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=61 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=57 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=58 for unequipped circuit (typ=RSC), link 'l2'. >>>> [Dec 15 12:55:29] WARNING[5540]: l4isup.c:4803 l4isup_event: Received >>>> CIC=59 for unequipped circuit (typ=RSC), link 'l2'. >>>> >>>> >>>> Kind regards >> >> >> >> -- >> _____________________________________________________________________ >> -- 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
