hello Claudio Aznar!
I guess you may feel luckier reading this post, as I was the one victim of
the same problem that you are currently in (problem: bringing the asterisk
server up on SS7 link with SIEMENS EWSD for successful calls, be it chan_ss7
or libss7). I tried both & had the same results as you have for each one, I
explained & posted the same as you did rather more than that including
exchange of posts with Mathew Fredrickson. Then, in an effort of leaving no
stone unturned, the solution came across & was lying in changing the CIC on
EWSD side from 1 to 0 while keeping firstcic=1 on asterisk (chan_ss7-0.8.4or
0.9). doing this made every thing working fine i.e. all channels became idle
& started receiving calls. Because the CIC mapping on the EWSD side starts
from CIC = 0 & remains the same for the whole trunk group i.e. E1, and for
the next trunkGroup/E1 there is another CIC, they represent it like this:
CIC=0-1
CIC=1-2
CIC=2-3
…
Where in CIC=0-1, the '0' specifies the CIC and the '1' is the trunkGroup/E1
number and similarly in CIC=1-2, the '1' is CIC & '2' is E1 number and so
on. So based on my experience, I can guess with a high probability that your
telco side has set CIC=1-1 on their EWSD as they generally do it. If on EWSD
you find it so, then ask the EWSD ppl to change their CIC to '0' for the
trunkGroup/E1 that is for your asterisk server connectivity and keep your
firstcic=1 & channels => 1-15,17-31 in the link settings in ss7.conf
I hope it'll work for you too
My posts when I was in the same prob are here:
http://lists.digium.com/pipermail/asterisk-ss7/2006-December/000928.html
http://lists.digium.com/pipermail/asterisk-ss7/2006-November/000716.html
On 6/30/07, Claudio Aznar <[EMAIL PROTECTED]> wrote:
Testing with chan_ss7 & Asterisk 1.4, I have INSERVICE with "ss7 link
status" and in the SDE side I look the link up too but don't work.
Like that libss7, the link go from DOWN to INSERVICE, I attach a little
debug from chan_ss7
sspVoIP*CLI> ss7 link status
linkset siuc, link l1, schannel 16, NOT_ALIGNED, rx: 0, tx: 4/4,
sentseq/lastack: 127/127, total 668544, 668592
sspVoIP*CLI> ss7 linestat 1
Linkset: siuc
CIC 2 Idle Reset pending BLOCKED Link down
CIC 3 Idle Reset pending BLOCKED Link down
....
CIC 31 Idle Reset pending BLOCKED Link down
And I see this WARNINGs & ERRORs
[Jun 29 19:21:35] WARNING[3930]: mtp.c:791 timeout_sltm_t1: No SLTA
received
within Q.707 timer T1, trying again on link 'l1'.
[Jun 29 19:21:44] ERROR[3930]: mtp.c:796 timeout_sltm_t1: No SLTA received
within Q.707 timer T1, faulting link on link 'l1'.
[Jun 29 19:21:44] WARNING[3930]: chan_ss7.c:637 monitor_main: MTP is now
DOWN on link 'l1'.
Regards,
Claudio Aznar
-----Mensaje original-----
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew
Fredrickson
Enviado el: Viernes, 29 de Junio de 2007 02:23 p.m.
Para: asterisk-ss7@lists.digium.com
Asunto: Re: [asterisk-ss7] Siemens SDE - libss7
On Jun 29, 2007, at 11:49 AM, Claudio Aznar wrote:
> Dear Matthew,
>
> Thank you for your answer. I'm attaching the config files so you
can
> see them.
> The only msg that I can view in the PBX side is "SLT fault".
> In the README os libss7 it says that it was tested with EWSD, do
you
> know the config files used for test?
You timing looks good. That's the biggest thing I was concerned
about. I can't remember exactly, it was either a TE4xxp or a TE110P,
when I tested it, but it was one of the two. Maybe finding further
into what that error means could help us. It's possible that it's
expecting periodic SLTM messages, which we don't do unless the other
end does first right now. But the meaning of that error is what
ultimately matters.
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
>
> Regards,
> Claudio Aznar
>
>
> -----Mensaje original-----
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Matthew
> Fredrickson
> Enviado el: Viernes, 29 de Junio de 2007 11:32 a.m.
> Para: asterisk-ss7@lists.digium.com
> Asunto: Re: [asterisk-ss7] Siemens SDE - libss7
>
> It almost looks like layer 1 problems. The other end is dropping the
> link for some reason (sending SIOS after the link has been aligned).
> Is your timing correct in your zaptel.conf?
>
> ---
> Matthew Fredrickson
> Software Engineer
> Digium, Inc.
>
> On Jun 28, 2007, at 7:47 PM, Claudio Aznar wrote:
>
>> Dear list,
>>
>>
>>
>> I'm trying to configure an Asterisk GW (Debian sarge w/
>> last trunk cvs version - r69519M). All the source compile fine and
>> I can install them without problem.
>>
>>
>>
>> In the Asterisk side, I see this:
>>
>>
>>
>> *CLI> ss7 show linkset 1
>>
>> SS7 linkset 1 status: Up
>>
>> *CLI> zap show status
>>
>> Description Alarms
>> IRQ bpviol CRC4
>>
>> Digium Wildcard TE110P T1/E1 Card 0 OK
>> 0 0 0
>>
>> *CLI> zap show channels
>>
>> Chan Extension Context Language MOH
>> Interpret Blocked State
>>
>> pseudo prepago es
>> default In Service
>>
>> 1 prepago es
>> default Out of Ser
>>
>> 2 prepago es
>> default Out of Ser
>>
>> .
>>
>> 31 prepago es
>> default Out of Ser
>>
>>
>>
>> In the PBX side (Siemens SDE, SS7 protocols like EWSD),
>> the status seems to be down.
>>
>> When a set debug for SS7, I see this:
>>
>>
>>
>> *CLI> ss7 debug linkset 1
>>
>> Enabled debugging on linkset 1
>>
>> *CLI> Len = 3 [ 81 82 00 ]
>>
>> FSN: 2 FIB 1
>>
>> BSN: 1 BIB 1
>>
>>> [0] FISU
>>
>>
>>
>> Len = 2 [ 82 81 ]
>>
>> FSN: 1 FIB 1
>>
>> BSN: 2 BIB 1
>>
>> <[0] LSSU SIOS
>>
>>
>>
>> Link state change: INSERVICE -> IDLE
>>
>> Len = 2 [ ff ff ]
>>
>> FSN: 127 FIB 1
>>
>> BSN: 127 BIB 1
>>
>> <[0] LSSU SIO
>>
>>
>>
>> Link state change: IDLE -> ALIGNED
>>
>> Len = 2 [ ff ff ]
>>
>> FSN: 127 FIB 1
>>
>> BSN: 127 BIB 1
>>
>> <[0] LSSU SIE
>>
>>
>>
>> Link state change: NOTALIGNED -> PROVING
>>
>> T4 expired!
>>
>> Link state change: PROVING -> ALIGNEDREADY
>>
>> Len = 3 [ ff ff 00 ]
>>
>> FSN: 127 FIB 1
>>
>> BSN: 127 BIB 1
>>
>>> [0] FISU
>>
>>
>>
>> Len = 1 [ ff ]
>>
>> FSN: 127 FIB 1
>>
>> BSN: 127 BIB 1
>>
>> <[0] FISU
>>
>>
>>
>> Link state change: ALIGNEDREADY -> INSERVICE
>>
>> Len = 20 [ ff 80 11 81 0c 72 83 0c 11 a0 32 35 36 34 32 38 36 32 38
>> 38 ]
>>
>> [ ff 80 11 ]
>>
>> FSN: 0 FIB 1
>>
>> BSN: 127 BIB 1
>>
>>> [0] MSU
>>
>> [ 81 ]
>>
>> Network Indicator: 2 Priority: 0 User Part: STD_TEST
>>
>> [ 0c 72 83 0c ]
>>
>> OPC 12813 DPC 12812 SLS 0
>>
>> [ 11 ]
>>
>> H0: 1 H1: 1
>>
>> Message type: COO
>>
>>
>>
>> Len = 21 [ ff 80 14 81 0d 32 83 1c 11 d0 03 01 54 45 53 54 5f 43 4f
>> 4e 46 ]
>>
>> [ ff 80 14 ]
>>
>> FSN: 0 FIB 1
>>
>> BSN: 127 BIB 1
>>
>> <[0] MSU
>>
>> [ 81 ]
>>
>> Network Indicator: 2 Priority: 0 User Part: STD_TEST
>>
>> [ 0d 32 83 1c ]
>>
>> OPC 12812 DPC 12813 SLS 1
>>
>> [ 11 ]
>>
>> H0: 1 H1: 1
>>
>> Message type: COO
>>
>>
>>
>> Len = 23 [ 80 81 14 81 0c 72 83 0c 21 d0 03 01 54 45 53 54 5f 43 4f
>> 4e 46 49 47 ]
>>
>> [ 80 81 14 ]
>>
>> FSN: 1 FIB 1
>>
>> BSN: 0 BIB 1
>>
>>> [0] MSU
>>
>> [ 81 ]
>>
>> Network Indicator: 2 Priority: 0 User Part: STD_TEST
>>
>> [ 0c 72 83 0c ]
>>
>> OPC 12813 DPC 12812 SLS 0
>>
>> [ 21 ]
>>
>> H0: 1 H1: 2
>>
>> Message type: COA
>>
>>
>>
>> I appreciate very much your help.
>>
>>
>>
>> Regards,
>>
>> Claudio Aznar
>>
>> _______________________________________________
>> --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
> <zaptel.conf>
> <zapata.conf>
> _______________________________________________
> --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
