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 <edd at edd.za.net> 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 <amish at 3g.co.za> 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<edd at edd.za.net> ?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 >> >
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