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>
|