Thank you guys for your insightful responses. My issue was indeed CIC mapping. It turns out that only the 1st E1 was enabled, after disabling all others and making sure E1 #1 is in the 1st slot, It worked like magic. I had gotten so dirsturbed, was about to insist they give me ISDN (PRI) lines (yet they only use ss7)..lol Anyway, thank you! I am now continuing with more stress testing. I have 8 E1s, 4 in each box, had wanted to do fail over using chan_ss7 but chan_ss7 didnt work for me. Thats what am now gonna try out again. Kind Regards, Tim 2010/11/29 Horacio J. Pe?a <horape at compendium.com.ar>: > 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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