Hi Bipin, Thanks for your reply.. Please check my log when the call was successfully answered.. ?== Using UDPTL CoS mark 5 ??? -- Executing [88029841234 at phones:1] Set("SIP/sysmaster-00000004", "CALLERID(ani)=9610999207") in new stack ??? -- Executing [88029841234 at phones:2] Set("SIP/sysmaster-00000004", "CALLERID(num)=9610999207") in new stack ??? -- Executing [88029841234 at phones:3] Dial("SIP/sysmaster-00000004", "DAHDI/G1/029841234") in new stack ??? -- Called G1/0029841234 Unhandled optional parameter 0x2d 'Unknown' [0x0 0x0 ] ??? -- DAHDI/62-1 answered SIP/sysmaster-00000004 ??? -- Hungup 'DAHDI/62-1' ? == Spawn extension (phones, 88029841234, 3) exited non-zero on 'SIP/sysmaster-00000004' ? Regards Saleh ________________________________ From: bipin singh <bipinraghuvanshi at gmail.com> To: rubel pool <rubel_ete at yahoo.com>; asterisk-ss7 at lists.digium.com Sent: Thursday, February 21, 2013 10:23 AM Subject: Re: No audio in ss7 link Hi pool, Check incoming IVR calls and also DTMF input., On Mon, Feb 18, 2013 at 4:51 PM, rubel pool <rubel_ete at yahoo.com> wrote: Dear all, > >In my ss7 link,? my link is up with all configuration.... My connectivity with carrier is 2 E1.. One E1 for signalling and another E1 for Voice.. > >so from last few years, my calling was ok but unfortunately my voice is stop... where as Carrier said they have nothing change in their system.. but where is the problem??.. > >My asterisk version 1.6.2.20 >?SS7 Card: Digium, Inc. Wildcard TE420P quad-span >? >## vi /etc/dahdi/system.conf > ># Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource >span=1,1,0,ccs,hdb3 >bchan=1-31 > ># Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS >span=2,2,0,ccs,hdb3,crc4 >mtp2=32 > > > >## vi /etc/asterisk/chan_dahdi.conf > >[channels] >language=en >context=from-btcl >switchtype=national >signalling=ss7 >toneduration=300 >usecallerid=yes >callwaiting=yes >usecallingpres=no >callwaitingcallerid=yes >threewaycalling=yes >transfer=yes >canpark=yes >cancallforward=yes >callreturn=yes >echocancel=yes >echocancelwhenbridged=no >group=1 >callgroup=1 >pickupgroup=1 >ss7type = itu >ss7_called_nai=international >ss7_calling_nai=national >relaxdtmf=yes >rxgain=0.0 >txgain=0.0 >linkset = 1 >pointcode = XXXX >adjpointcode = XXXX >defaultdpc = XXXX >networkindicator=national >faxbuffers => 12,half >faxdetect=both >cicbeginswith=1 >channel =1-31 >sigchan=32 >;ss7_internationalprefix=00 >;ss7_nationalprefix=0 > >? > >where is the problem?? I matched the CIC with Carrier.. all is ok.. Please guide me.. > >Regards > >Saleh >-- >_____________________________________________________________________ >-- 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 bipinraghuvanshi at gmail.com bipin.singh at ehorizons.in -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130223/92302469/attachment.htm>
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