[asterisk-ss7] No Audio

Trevor Francis trevor.francis at tgrahamcapital.com
Tue Jul 12 09:06:35 CDT 2011 

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 <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
>>>>> 
>>>>> --
>>>>> 
>>>>> 
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>>>>> 
>>>>> -- 
>>>>> Robert
>>>>> 
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