Are you test only whith a dialout? Are you test with a inbound call, using a playback o read command? I had the same issue with asterisk 1.8 (i've tried with asterisk-1.8.3.3 and asterisk-1.8.4.2) when was a outbound call they don't pass a voice or dtmf but in a inbound call yes. I've tried with asterisk-1.6.2.18 and both type of call was OK. Regards, Val 2011/7/12 Abdul Basit <basit.engg at gmail.com> > have you tested with dahdi_monitor on the active channel? > > See if what audio side (Rx or Tx) you are getting. This is CIC miss-match > issue. > dahdi_monitor might help you figuring out the next CIC that has audio > channel. see all one by one. > > Also do it step by step. Stop all E1s as suggested and then start up in > steps. Monitor CIC with dahdi_monitor. > > > -- > Regards, > > Abdul Basit > > > > > On Tue, Jul 12, 2011 at 2:42 PM, Yo - <yoherman at gmail.com> wrote: > >> as my experience. Gtalk with telco. Shutdown all E1 port. Startup step by >> step, one by one E1 port. contact and contact telco to make sure cic match >> on right e1. >> >> >> On Tue, Jul 12, 2011 at 3:44 PM, Trevor Francis < >> trevor.francis at tgrahamcapital.com> wrote: >> >>> So for 4 E1s I would do this? >>> >>> mtp2=1 >>> sigchan=1 >>> context=default >>> cicbeginswith = 1 >>> channel = 2-31 >>> cicbeginswith = 33 >>> channel = 32-62 >>> cicbeginswith = 65 >>> channel = 63-93 >>> cicbeginswith = 97 >>> channel = 94-124 >>> >>> >>> -- >>> Trevor G. Francis >>> Managing Member >>> trevor.francis at tgrahamcapital.com >>> >>> Ph. +1 405.445.4020 >>> Fx. +1 405.445.4021 >>> P.O Box 54771 >>> Oklahoma City, OK 73154 >>> MSN: trevor.francis at fiberhaus.com >>> Personal emails should be addressed to: tfrancis at fas.harvard.edu >>> -- >>> >>> On Jul 12, 2011, at 3:39 AM, Robert Thomas wrote: >>> >>> It's odd an start with 2 as the CIC number... I have never seen this at >>> least. Most of the time they are consecutive >>> >>> On Tue, Jul 12, 2011 at 3:37 AM, Trevor Francis < >>> trevor.francis at tgrahamcapital.com> wrote: >>> >>>> Its a Huawei switch. Any idea on what they standardize on as far as >>>> CICs? >>>> >>>> >>>> -- >>>> >>>> On Jul 12, 2011, at 3:34 AM, Robert Thomas wrote: >>>> >>>> The fact that you start using voice circuit #2m doesnt necesarily means >>>> they start counting from CIC #2. >>>> >>>> They could start CIC 1, in channel 2 and always be off by 1. You can try >>>> configuring with cicbegins with 1. >>>> >>>> On Tue, Jul 12, 2011 at 3:31 AM, Trevor Francis < >>>> trevor.francis at tgrahamcapital.com> wrote: >>>> >>>>> I have been told by the telco the following >>>>> >>>>> SLC= 0 >>>>> Signaling link = TS1 on 1st E1 >>>>> Voice Circuits = 2 - 31, 33-63, 65-95, 97-127 >>>>> >>>>> What else am I missing? >>>>> -- >>>>> >>>>> On Jul 12, 2011, at 3:26 AM, Robert Thomas wrote: >>>>> >>>>> So you have the D channels Aligned and the LSSU go in both direction. >>>>> That does not guarantee the CIC are aligned. >>>>> >>>>> On Tue, Jul 12, 2011 at 3:25 AM, Trevor Francis < >>>>> trevor.francis at tgrahamcapital.com> wrote: >>>>> >>>>>> MTP2 link up (SLC 0) >>>>>> --- SS7 Up --- >>>>>> Resetting CICs 2 to 31 >>>>>> Resetting CICs 33 to 63 >>>>>> Resetting CICs 65 to 95 >>>>>> Resetting CICs 97 to 127 >>>>>> Got reset acknowledgement from CIC 2 to 31. >>>>>> Got reset acknowledgement from CIC 33 to 63. >>>>>> Got reset acknowledgement from CIC 65 to 95. >>>>>> Got reset acknowledgement from CIC 97 to 127. >>>>>> >>>>>> They are talking to each other.... >>>>>> >>>>>> -- >>>>>> Trevor G. Francis >>>>>> Managing Member >>>>>> trevor.francis at tgrahamcapital.com >>>>>> >>>>>> Ph. +1 405.445.4020 >>>>>> Fx. +1 405.445.4021 >>>>>> P.O Box 54771 >>>>>> Oklahoma City, OK 73154 >>>>>> MSN: trevor.francis at fiberhaus.com >>>>>> Personal emails should be addressed to: tfrancis at fas.harvard.edu >>>>>> -- >>>>>> >>>>>> On Jul 12, 2011, at 3:19 AM, James zhu wrote: >>>>>> >>>>>> hi: >>>>>> yes, it should be a problem with CIC mismatched. >>>>>> >>>>>> Best regards, >>>>>> James.zhu >>>>>> Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, >>>>>> gateway(fxs/fxo/pri<->SIP). >>>>>> website: www.voipviews.com >>>>>> >>>>>> >>>>>> ------------------------------ >>>>>> Date: Tue, 12 Jul 2011 03:17:22 -0500 >>>>>> From: thomcr at gmail.com >>>>>> To: asterisk-ss7 at lists.digium.com >>>>>> Subject: Re: [asterisk-ss7] No Audio >>>>>> >>>>>> How do you know you have your CICs aligned? >>>>>> >>>>>> You and the TELCO could start counting from the same place, however >>>>>> the E1 may be crossed. This happend to me when 2nd E1 of the TELCO was the >>>>>> 3rd for me. The cal would be established on CIC 33 for Example on E1 #2, >>>>>> but my server was reciving it on #3. >>>>>> >>>>>> I would recommend you to disconnect all your E1 and confirm with the >>>>>> alarms the TELCO has them on the same order than you. Or just try the >>>>>> different combination. >>>>>> >>>>>> As well double check your CIC count to make sure it matched the TELCO. >>>>>> >>>>>> On Tue, Jul 12, 2011 at 3:08 AM, Trevor Francis < >>>>>> trevor.francis at tgrahamcapital.com> wrote: >>>>>> >>>>>> We have gone round and round on getting our ss7 link up. We can get >>>>>> the cics to align and the signaling link to come up. However, when we dial >>>>>> there is no audio in either direction. >>>>>> >>>>>> Chan_dahdi: >>>>>> >>>>>> >>>>>> [trunkgroups] >>>>>> [channels] >>>>>> context=default >>>>>> usecallerid=yes >>>>>> hidecallerid=no >>>>>> callwaiting=no >>>>>> usecallingpres=yes >>>>>> threewaycalling=no >>>>>> transfer=yes >>>>>> canpark=no >>>>>> cancallforward=no >>>>>> callreturn=no >>>>>> echocancel=yes >>>>>> echocancelwhenbridged=yes >>>>>> relaxdtmf=yes >>>>>> rxgain=0.0 >>>>>> txgain=0.0 >>>>>> immediate=no >>>>>> prematureaudio=no >>>>>> language=en >>>>>> group=1 >>>>>> signalling = ss7 >>>>>> ss7type = itu >>>>>> >>>>>> >>>>>> linkset = 1 >>>>>> pointcode=6314 ; switch point code >>>>>> adjpointcode=12450 ; peer point code. >>>>>> defaultdpc=12450 ; per point code. >>>>>> networkindicator=international >>>>>> slc=0 >>>>>> ;ss7_internationalprefix = 00 >>>>>> ;ss7_nationalprefix = 0 >>>>>> ;ss7_subscriberprefix = >>>>>> ;ss7_unknownprefix = >>>>>> >>>>>> mtp2=1 >>>>>> sigchan=1 >>>>>> context=default >>>>>> cicbeginswith = 2 >>>>>> channel = 2-31 >>>>>> cicbeginswith = 33 >>>>>> channel = 32-62 >>>>>> cicbeginswith = 65 >>>>>> channel = 63-93 >>>>>> cicbeginswith = 97 >>>>>> channel = 94-124 >>>>>> >>>>>> Dahdi system.conf >>>>>> >>>>>> span=1,1,0,ccs,hdb3 >>>>>> bchan=2-31 >>>>>> dchan=1 >>>>>> echocanceller=mg2,2-31 >>>>>> >>>>>> span=2,0,0,ccs,hdb3 >>>>>> bchan=32-62 >>>>>> echocanceller=mg2,32-62 >>>>>> >>>>>> span=3,0,0,ccs,hdb3 >>>>>> bchan=63-93 >>>>>> echocanceller=mg2,63-93 >>>>>> >>>>>> span=4,0,0,ccs,hdb3 >>>>>> bchan=94-124 >>>>>> echocanceller=mg2,94-124 >>>>>> >>>>>> loadzone = fr >>>>>> defaultzone = fr >>>>>> >>>>>> >>>>>> Any ideas? >>>>>> >>>>>> Running Asterisk 1.8.4.4, DAHDI Version: 2.4.1.2 Echo Canceller: MG2, >>>>>> libss7 version: 1.0.2 >>>>>> >>>>>> -- >>>>>> >>>>>> >>>>>> -- >>>>>> _____________________________________________________________________ >>>>>> -- 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Robert >>>>>> >>>>>> -- >>>>>> _____________________________________________________________________ -- >>>>>> 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 >>>>>> -- >>>>>> _____________________________________________________________________ >>>>>> -- 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 >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> _____________________________________________________________________ >>>>>> -- 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Robert >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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 >>>>> >>>>> >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Robert >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> >>> -- >>> Robert >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 > > > -- > _____________________________________________________________________ > -- 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... 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