[asterisk-ss7] chan_ss7 error - Write buffer full on CIC

[Gregory Massel]

Hi

I don't think your suggestion of moving to libss7 is practical. I'm specifically using chan_ss7 rather than libss7 because the carrier we're interconnecting to runs two STPs in a "mated pair" configuration and, to the best of my knowlege, libss7 does not support that.

The chan_ss7 config looks as follows:

[linkset-ls1]
...
combined => mated-pair

[linkset-ls2]
...
combined => mated-pair

[link-l1]
linkset => ls1
stp => 123
...

[link-l2]
linkset => ls1
stp => 123
...

[link-l3]
linkset => ls2
stp => 456
...

[host-abc]
default_linkset => ls1
opc => 776
dpc => ls1:1001
dpc => ls2:2002
...



The key thing about this config is that we need to be able to receive signalling on any of the links from either stp (123 or 456) and to be able to send signalling to either stp via any of the links. To complicate matters, the adjacent point codes are different to the STP point codes.

This is a rather complex setup, however, it's not at our discretion to change it; that's how the other telco works. Chan_ss7 1.3 handles this fairly well. The only problem is that is cannot handle the same CIC numbers being used on different links within different linksets that are in a mated pair (it seemed to get confused when group reset messages came through), however, the remote telco was willing to use different CIC numbers to work around that.

I've looked at the libss7 documentation extensively and I cannot find anything to suggest that it can handle the mated pair as above.

--Greg
  ----- Original Message ----- 
  From: bipin singh 
  To: asterisk-ss7 at lists.digium.com 
  Sent: Wednesday, January 27, 2010 6:50 AM
  Subject: Re: [asterisk-ss7] chan_ss7 error - Write buffer full on CIC


  hi
             use libss7 its work better than chan_ss7


  On Wed, Jan 27, 2010 at 5:15 AM, Gregory Massel <greg at csurf.co.za> wrote:

    > from chan_ss7 faq:
    [snip]

    > you can try:
    > - newer version of chan_ss7
    > - jitter buffer option in ss7.conf
    > [jitter]
    > jbenable = yes
    > jbmaxsize = 1000
    > jbresyncthreshold = 1000
    > jbimpl = adaptive


    Thank you!

    Apologies for asking a question in the FAQ. One more question: where do I
    find the latest FAQ? It doesn't appear to be included in the
    chan_ss7-1.3.tar.gz source and the FAQ on
    http://www.voip-info.org/wiki/view/Asterisk+ss7+faq doesn't cover this
    issue.

    Thanks
    Greg



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  BIPIN RAGHUVANSHI
  OPERATION HEAD
  ASTERISK (DEVELOPMENT AND RESEARCH)  
  WWW.EHORIZONS.IN
  011-32323262



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