On ISUP each channel is identified by a number called CIC, and the assignation CIC <-> E1 timeslot is arbitrary. Your CIC 3 can as easily be channel 3 of your first E1 as channel 27 of your 2nd E1 (I think it could even be a copper line or a RTP stream, but let?s not go there) If you establish a call on CIC 3 and you believe CIC 3 is the channel 3 of your first E1 and the other side believes it?s the channel 4 of that same E1, you'll get silence. An easy test to see if there is a problem with the CICs is making several calls at once. If you get audio, but from the wrong person it's surely a CIC mismatch, and you can easily discover the right config. Say that you do the calls: A -> B, CIC 2 C -> D, CIC 3 E -> F, CIC 4 and when the calls are connected A is talking to B, that means that the channel you believe to be CIC 2 is really CIC 3 (That would be the outcome if the other side numbers the CIC from 1 as somebody suggested) On Mon, Nov 29, 2010 at 06:48:44PM +0300, Timothy Smith wrote: > Thank Dave do your advise. > > Please advise me further, how do I verify the CICs and T1-1 (do u mean > time slots?) are line up correctly? I can meet the telco engineer but > need to explain to him properly and make my point. Unfornately, he > doesnt know asterisk :( (only knows his huawei switch). > > By the way, i forgot to mention, when I turn on crc4 (in my > dahdi/system.conf), the link just starts coming up and down every > time! (see output below) > > [Nov 29 10:47:45] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2 > link down (SLC 0) > MTP2 link up (SLC 0) > [Nov 29 10:47:48] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2 > link down (SLC 0) > MTP2 link up (SLC 0) > [Nov 29 10:47:50] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2 > link down (SLC 0) > MTP2 link up (SLC 0) > Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting > retransmission > MSU received, though still waiting for retransmission start. Dropping. > Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting > retransmission > > > Thanks, > Tim > > > On Mon, Nov 29, 2010 at 6:40 PM, dave george <dgeorge at teletoneinc.com> wrote: > > Hi Tim, > > > > Make sure your CICs line up. ?Check that your T1-1 is the other side T1-1. > > I had a similar problem and my CIC was not lined up. ?My T1-1 was their > > T1-5. > > > > > > Thanks, > > Dave > > > > -----Original Message----- > > From: asterisk-ss7-bounces at lists.digium.com > > [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Timothy Smith > > Sent: Monday, November 29, 2010 9:57 AM > > To: asterisk-ss7 at lists.digium.com > > Subject: Re: [asterisk-ss7] Help with SS7 (No Audio) > > > > Thank you Gentlemen for your responses. > > > > I have done the dahdi_monitor, its only TX that has some input (see > > sample output below). Thats for both outgoing and incoming calls. > > > > How can I verify the circuit mapping? My core engineer (telco company) > > said that he is using the 1st channel for signalling and the rest for > > voice. > > > > I appreciate your help. > > > > Tim > > > > [root at ivr asterisk]# dahdi_monitor 12 -vvv > > > > Visual Audio Levels. > > -------------------- > > ?Use chan_dahdi.conf file to adjust the gains if needed. > > > > ( # = Audio Level ?* = Max Audio Hit ) > > <----------------(RX)----------------> > > <----------------(TX)----------------> > > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?################### ?* > > ? ? ^Ccntrl-c pressed 0) Tx: ?2516 ( 3960) > > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?################# ? ?* > > ? ? ? ?Rx: ? ? 0 ( ? ?0) Tx: ?3308 ( 3960)done cleaning up ... > > exiting. > > [root at ivr asterisk]# dahdi_monitor 3 -vvv > > > > Visual Audio Levels. > > -------------------- > > ?Use chan_dahdi.conf file to adjust the gains if needed. > > > > ( # = Audio Level ?* = Max Audio Hit ) > > <----------------(RX)----------------> > > <----------------(TX)----------------> > > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?########### ? ?* > > ? ? ^Ccntrl-c pressed 0) Tx: ?2111 ( 2790) > > ? Rx: ? ? 0 ( ? ?0) Tx: ?2035 ( 2790)done cleaning up ... exiting. > > [root at ivr asterisk]# > > > > > > On Mon, Nov 29, 2010 at 4:57 PM, Abdul Basit <basit.engg at gmail.com> wrote: > >> Try sending a call via call file and see if you are getting both call > > legs. > >> callchannel.sh > >> #!/bin/bash > >> echo "Channel: DAHDI/$1/$2 > >> Callerid: $2 > >> MaxRetries: 2 > >> RetryTime: 60 > >> WaitTime: 30 > >> Context: ss7 > >> Application: Echo"?> /var/spool/asterisk/tmp/test.call > >> mv /var/spool/asterisk/tmp/test.call /var/spool/asterisk/outgoing > >> dahdi_monitor $1 -vv > >> This is the way i verify the call legs. > >> chmod +x callchannel.sh > >> ./callchannel.sh channelNumber someNumber > >> ./callchannel.sh 3 123456789 > >> > >> Most of the time problem is cic miss-match. > >> I hope this will help you debugging the issue. > >> > >> > >> On Mon, Nov 29, 2010 at 6:34 PM, Timothy Smith <timotsmith at gmail.com> > > wrote: > >>> > >>> Dear Users, > >>> > >>> I seeking help on with the asterisk+libss7. ?the call is successfully > >>> setup but no audio either end. > >>> > >>> I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2, > >>> chan_dahdi.c is too bing but i can send it if required(perhaps to add > >>> p->dialing = 0. I didnt do it > >>> correctly?) > >>> > >>> I appreciate your help in advance. Could someone please send me > >>> working confs/chan_dahdi.conf please! > >>> > >>> [root at ivr asterisk]# cat chan_dahdi.conf > >>> [trunkgroups] > >>> [channels] > >>> echocancel=yes > >>> echocancelwhenbridged=yes > >>> group=1 > >>> signalling=ss7 > >>> ss7type=itu > >>> ss7_called_nai=national > >>> ss7_calling_nai=national > >>> linkset=1 > >>> pointcode=25 > >>> adjpointcode=33 > >>> defaultdpc=33 > >>> networkindicator=national > >>> sigchan=1 > >>> cicbeginswith=2 > >>> channel=2-124 > >>> ss7_internationalprefix=000 > >>> ss7_nationalprefix=0 > >>> context=ss7 > >>> [root at ivr1 asterisk]# cat /etc/dahdi/system.conf > >>> span=1,1,0,ccs,hdb3 > >>> bchan=2-31 > >>> mtp2=1 > >>> span=2,2,0,ccs,hdb3 > >>> bchan=32-62 > >>> span=3,3,0,ccs,hdb3 > >>> bchan=63-93 > >>> span=4,4,0,ccs,hdb3 > >>> bchan=94-124 > >>> > >>> loadzone ? ? ? ?= us > >>> defaultzone ? ? = us > >>> [root at ivr asterisk]# > >>> > >>> > >>> Thank you! > >>> Kind Regards, > > -- > _____________________________________________________________________ > -- 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 -- Horacio J. Pe?a horape at compendium.com.ar horape at uninet.edu
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