[asterisk-ss7] No audio using libss7 over E1

[David Wilson]

Hi Krzysztof,

Thank you for your reply. It is greatly appreciated!

Please find my responses below.

 >Alignment on link does not mean that - you start counting CICs from 
"1" and your partner does the same (one is one on both sides, some telco 
switches count 1st >CIC as 0, or even 2).
Yes. That makes sense.


 >Here you say, that first available channel has logical number one, and
 >you try to make 1st call on CIC 1, so it looks good.
 >Maybe simply - you should try to disable one E1 link? Simply comment
 >out - the second E1 and see if it's not mis-cabling ?

The system is in another African country so instead of me physically 
pulling out the E1 cable from the system I tested by 'blocking' certain 
CICs in my linkset1 by using the 'ss7 block cic 1 [channel]' command. 
The provider said that he could see the corresponding blocked CICs and 
they appeared to match up to his CICs. Unfortunately without me being at 
the same physical location I am unable to view the provider's equipment 
and logs to verify that the CICs are actually matching on his side.

 >If the same "no audio" happen on the only one digital link (one E1) -
 >it's in 99.999% - mis numbered CIC - have you access to any passive
 >"recording" equipment ?
 >Another Sangoma card with at least 2 ports could allow for listening
 >if audio is being put on any timeslot (CIC, timeslot, Bchannel - call
 >it as you wish).
Unfortunately I don't have access to recording equipment or another 
Sangoma card.

Something I have picked up in my Asterisk log is the following:
/[Aug  1 20:01:21] VERBOSE[30284] chan_dahdi.c: Resetting CIC 47
[Aug  1 20:01:21] WARNING[30284] chan_dahdi.c: RSC on unconfigured CIC 47
[Aug  1 20:01:51] VERBOSE[30284] chan_dahdi.c: Resetting CIC 63
[Aug  1 20:01:51] WARNING[30284] chan_dahdi.c: RSC on unconfigured CIC 63/


Could this possibly confirm the CIC alignment problem that we suspect? I 
don't have a bchan/CIC configured for 47 or 63
Applicable piece from chan_dahdi.conf:
;First E1
       sigchan=16
       cicbeginswith=1
       channel=1-15
       cicbeginswith=17
       channel=17-31

;Second E1
       sigchan=47
       cicbeginswith=32
       channel=32-46
       cicbeginswith=48
       channel=48-62


Thank you for your assistance so far!



Get important Linux and industry-related news at: facebook.com/dcdata 
<http://facebook.com/dcdata>

Kind regards,

David Wilson
CNS,CLS, LINUX+, CLA, DCTS, LPIC3
*LinuxTech CC t/a DcData*
CK number: 2001/058368/23
*Website:* 	http://www.dcdata.co.za
*Support:* 	+27(0)860-1-LINUX
*Mobile:* 	+27(0)824147413
*Tel:* 	+27(0)333446100
*Fax:* 	+27(0)866878971


On 08/01/2012 06:41 PM, Krzysztof Drewicz wrote:
> 2012/8/1 David Wilson<dave at dcdata.co.za>
>> With debugging enabled on my SS7 linkset I see the following in my logs. Unfortunately not being familiar with SS7 it doesn't mean too much to me. Everything that I've read indicates that a "no audio" problem with SS7 generally indicates misaligned CICs but according to the operator they are reported to be aligned ok.
>
> Alignment on link does not mean that - you start counting CICs from
> "1" and your partner does the same (one is one on both sides, some
> telco switches count 1st CIC as 0, or even 2).
>
>>                  CIC: 1
>> ;Sangoma A102 port 1 [slot:4 bus:12 span:1]<wanpipe1>
> So - it should be 1st CIC
>
>> ;First E1
>> sigchan=16
>> cicbeginswith=1
>> channel=1-15
> Here you say, that first available channel has logical number one, and
> you try to make 1st call on CIC 1, so it looks good.
> Maybe simply - you should try to disable one E1 link? Simply comment
> out - the second E1 and see if it's not mis-cabling ?
> If the same "no audio" happen on the only one digital link (one E1) -
> it's in 99.999% - mis numbered CIC - have you access to any passive
> "recording" equipment ?
> Another Sangoma card with at least 2 ports could allow for listening
> if audio is being put on any timeslot (CIC, timeslot, Bchannel - call
> it as you wish).
>
> --
> _____________________________________________________________________
> -- 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/20120801/089eda9b/attachment.htm>