The problem was due to "r" being enabled in the dialstring. Seems there is a bug in the ringer as after faking ringback it is unable to bridge channels properly.
-- Trevor G. Francis -- On Jul 12, 2011, at 8:29 AM, Val Appleyard wrote: > 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 <[email protected]> > 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 - <[email protected]> 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 > <[email protected]> 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 > [email protected] > > Ph. +1 405.445.4020 > Fx. +1 405.445.4021 > P.O Box 54771 > Oklahoma City, OK 73154 > MSN: [email protected] > Personal emails should be addressed to: [email protected] > -- > > 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 >> <[email protected]> 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 >>> <[email protected]> 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 >>>> <[email protected]> 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 >>>> [email protected] >>>> >>>> Ph. +1 405.445.4020 >>>> Fx. +1 405.445.4021 >>>> P.O Box 54771 >>>> Oklahoma City, OK 73154 >>>> MSN: [email protected] >>>> Personal emails should be addressed to: [email protected] >>>> -- >>>> >>>> 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: [email protected] >>>>> To: [email protected] >>>>> 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 >>>>> <[email protected]> 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 > > -- > _____________________________________________________________________ > -- 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
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