[asterisk-ss7] Issue with Interconnect
Edrich de Lange
edd at edd.za.net
Fri Dec 17 07:19:18 UTC 2010
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
>>
>>
>>
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