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 -- _____________________________________________________________________ -- 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 -- BIPIN RAGHUVANSHI OPERATION HEAD ASTERISK (DEVELOPMENT AND RESEARCH) WWW.EHORIZONS.IN 011-32323262 ------------------------------------------------------------------------------ -- _____________________________________________________________________ -- 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/20100128/2d0905d3/attachment.htm