[asterisk-ss7] asterisk-ss7 Digest, Vol 61, Issue 35

Ziad Salameh ziad at doubleu.mobi
Fri Mar 26 04:33:22 CDT 2010 

Dear All,

I am not sure if I can post this here or I should start a new thread but
here goes...
I setup Libss7 with asterisk 1.6 , linkset is UP, I can make successful
outbound calls , I can hear the other party well.
Now the odd thing is that when someone tries an inbound call , they hear
nothing I tried setting up MOH, I also tried to record the incoming call but
I hear nothing.
Is anyone facing such an issue, will you please help me shed some light on
this issue.

Thank you,
Ziad

-----Original Message-----
From: asterisk-ss7-bounces at lists.digium.com
[mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of
asterisk-ss7-request at lists.digium.com
Sent: Friday, March 26, 2010 9:09 AM
To: asterisk-ss7 at lists.digium.com
Subject: asterisk-ss7 Digest, Vol 61, Issue 35

Send asterisk-ss7 mailing list submissions to
	asterisk-ss7 at lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.digium.com/mailman/listinfo/asterisk-ss7
or, via email, send a message with subject or body 'help' to
	asterisk-ss7-request at lists.digium.com

You can reach the person managing the list at
	asterisk-ss7-owner at lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. help (paul marcovici)


----------------------------------------------------------------------

Message: 1
Date: Fri, 26 Mar 2010 00:08:30 -0700 (PDT)
From: paul marcovici <paul.marcovici at yahoo.com>
Subject: [asterisk-ss7] help
To: asterisk-ss7 at lists.digium.com
Message-ID: <966245.47373.qm at web113601.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="us-ascii"






________________________________
From: "asterisk-ss7-request at lists.digium.com"
<asterisk-ss7-request at lists.digium.com>
To: asterisk-ss7 at lists.digium.com
Sent: Fri, March 26, 2010 8:27:57 AM
Subject: asterisk-ss7 Digest, Vol 61, Issue 34

Send asterisk-ss7 mailing list submissions to
    asterisk-ss7 at lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
    http://lists.digium.com/mailman/listinfo/asterisk-ss7
or, via email, send a message with subject or body 'help' to
    asterisk-ss7-request at lists.digium.com

You can reach the person managing the list at
    asterisk-ss7-owner at lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. Re: IAM-REL instead of IAM-ACM-REL (Bruno Rodrigues de Mello)
   2. Re: IAM-REL instead of IAM-ACM-REL (Domjan Attila)
   3. Where's Matt been?  Well, here's the explanation
      (Matthew Fredrickson)
   4. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)
   5. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)


----------------------------------------------------------------------

Message: 1
Date: Thu, 25 Mar 2010 15:07:30 -0300
From: "Bruno Rodrigues de Mello" <shotsbros at hotmail.com>
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: <asterisk-ss7 at lists.digium.com>
Message-ID: <SNT127-DS14B34BD7A59DD4759F3770B0240 at phx.gbl>
Content-Type: text/plain; format=flowed; charset="utf-8";
    reply-type=original

Hi Attila, your patch more one time work without problems thank you again.

Regarding this email let me check other problem with you. I have a asterisk 
box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN to 
CISCO

TDM<----SS7--->ASTERISK<---ISDN---> CISCO

The call come from SS7 side and asterisk forward this call to ISDN. The 
cisco gateway plays a message in PROCEEDING. This message ask the user to 
put some digits. The calling side listen the message but when he put the 
digits the ISDN side don't receive the DTMF.

I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I can 
listen the audio and the dtmf and in ISDN side I don't listen the DTMF only 
the audio

I don't know why asterisk don't foward the audio received on SS7 side before

the ANM  during the early media.

Do you know anything about this ?


Regards,
Bruno Rodrigues

--------------------------------------------------
From: "Attila Domjan" <adomjan at tvnet.hu>
Sent: Wednesday, March 17, 2010 9:17 AM
To: <asterisk-ss7 at lists.digium.com>
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL

> -- 
> _____________________________________________________________________
> -- 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 




------------------------------

Message: 2
Date: Thu, 25 Mar 2010 23:47:30 +0100
From: Domjan Attila <adomjan at tvnet.hu>
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: asterisk-ss7 at lists.digium.com
Message-ID: <1269557250.2046.1.camel at localhost>
Content-Type: text/plain; charset="utf-8"

Hi,
I'm not sure it should work... it is not ss7 it is dahdi issue....

On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:
> Hi Attila, your patch more one time work without problems thank you again.
> 
> Regarding this email let me check other problem with you. I have a
asterisk 
> box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN to

> CISCO
> 
> TDM<----SS7--->ASTERISK<---ISDN---> CISCO
> 
> The call come from SS7 side and asterisk forward this call to ISDN. The 
> cisco gateway plays a message in PROCEEDING. This message ask the user to 
> put some digits. The calling side listen the message but when he put the 
> digits the ISDN side don't receive the DTMF.
> 
> I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I
can 
> listen the audio and the dtmf and in ISDN side I don't listen the DTMF
only 
> the audio
> 
> I don't know why asterisk don't foward the audio received on SS7 side
before 
> the ANM  during the early media.
> 
> Do you know anything about this ?
> 
> 
> Regards,
> Bruno Rodrigues
> 
> --------------------------------------------------
> From: "Attila Domjan" <adomjan at tvnet.hu>
> Sent: Wednesday, March 17, 2010 9:17 AM
> To: <asterisk-ss7 at lists.digium.com>
> Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
> 
> > -- 
> > _____________________________________________________________________
> > -- 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 --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100325/30e7c1e7
/attachment-0001.pgp 

------------------------------

Message: 3
Date: Thu, 25 Mar 2010 23:40:29 -0500
From: Matthew Fredrickson <creslin at digium.com>
Subject: [asterisk-ss7] Where's Matt been?  Well, here's the
    explanation
To: asterisk-ss7 at lists.digium.com
Message-ID: <4BAC3ABD.4000701 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hey all,

It's been a long time.  I apologize for my quietness for the last while 
here, it has been a very busy year this last year with some of the other 
projects I've been working on.

I just wanted to let everyone know that I'm still alive, and haven't 
given up or forgotten about libss7, and Asterisk with SS7.

I appreciate very much Attila's great work on getting libss7 polished 
up.  He's done a very good job taking it to the next level, and has done 
a great job helping everyone out.  Thanks very much Attila.

As far as getting his code merged back in, I was in the midst of this a 
while ago, but had to stop due to personal time issues as well as some 
other potentially architectural issues that subsequently came up. 
Hopefully we can get that moving again in the next little bit though.

I actually have been quite busy, and hopefully have some things to show 
for it, two things actually.

I'll not go into too much detail tonight (as it's getting quite late) 
but I'm looking for some hungry testers, that wouldn't mind beating up 
on some probably alpha level code.

They are:

1.) libss7 point code clustering support.

Basically, you can have Asterisk boxes share signalling links now using 
this code.  Although the signalling links are physically terminated on 
other machines, you can plug bearer T1s/E1s into other Asterisk boxes 
and virtually utilize the signalling links of the other boxes.

2.) A new channel driver, called chan_ccs, that allows, among other 
things, you to control MGCP media gateways for bearer trunks, instead of 
having to locally terminate them on the asterisk box that's controlling 
the signalling links.  There is also code in the same branch that has 
chan_ccs that modified chan_mgcp so that Asterisk can act as a media 
gateway (since I don't have any good real media gateways to test on). 
This basically means you can have Asterisk TDM channel scalability up to 
(in the ideal state) the same level as you can do with SIP with no 
media, per box.

In essence, this is turning Asterisk into a "true" softswitch, allowing 
native bridging between media gateways and any other RTP endpoint 
(including other media gateways).  This also means that you don't have 
to terminate bearer T1s/E1s on the main signalling box.

-- So, what does this mean for you, you may be asking?

These are really the next steps in making big SS7 work with Asterisk. 
They both allow for scaling a point code across multiple asterisk 
machines, and distribution of bearers on different boxes than the ones 
that contain signalling links.

Like I said though, most of the work is in a functional but early phase, 
and so I need some people that are interested enough in the added 
functionality that they're willing to work with potential hickups along 
the way.

Some of the changes I've had to make to libss7 have made it further more 
difficult to merge Attila's changes back in, which is the other reason 
why it has been so long and it still has not been merged.

If you're interested, either reply to me or this thread and let me know.

Thanks again,

Matthew Fredrickson
Digium, Inc.




------------------------------

Message: 4
Date: Fri, 26 Mar 2010 11:09:48 +0530
From: Anil Gupta <anilgupta83 at gmail.com>
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: asterisk-ss7 at lists.digium.com
Message-ID:
    <3bb51fcb1003252239s6542a998w21f6105261190a5f at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

I doubt ss7 starts transmitting audio because I can see asterisk does not
actually include optional backward call inband information parameter, inband
information parameter should be set to 1 for audio path to open before ANM.
I believe some switches could be configured to force this behavior. You
might want to record on a dahdi channel to make sure if audio is being
transmitted after ACM

Regards,

Anil

PS : Its will be better if you start a new thread for this.

On Fri, Mar 26, 2010 at 4:17 AM, Domjan Attila <adomjan at tvnet.hu> wrote:

> Hi,
> I'm not sure it should work... it is not ss7 it is dahdi issue....
>
> On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:
> > Hi Attila, your patch more one time work without problems thank you
> again.
> >
> > Regarding this email let me check other problem with you. I have a
> asterisk
> > box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN
> to
> > CISCO
> >
> > TDM<----SS7--->ASTERISK<---ISDN---> CISCO
> >
> > The call come from SS7 side and asterisk forward this call to ISDN. The
> > cisco gateway plays a message in PROCEEDING. This message ask the user
to
> > put some digits. The calling side listen the message but when he put the
> > digits the ISDN side don't receive the DTMF.
> >
> > I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I
> can
> > listen the audio and the dtmf and in ISDN side I don't listen the DTMF
> only
> > the audio
> >
> > I don't know why asterisk don't foward the audio received on SS7 side
> before
> > the ANM  during the early media.
> >
> > Do you know anything about this ?
> >
> >
> > Regards,
> > Bruno Rodrigues
> >
> > --------------------------------------------------
> > From: "Attila Domjan" <adomjan at tvnet.hu>
> > Sent: Wednesday, March 17, 2010 9:17 AM
> > To: <asterisk-ss7 at lists.digium.com>
> > Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
> >
> > > --
> > > _____________________________________________________________________
> > > -- 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...
URL:
http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/5ad6756d
/attachment-0001.htm 

------------------------------

Message: 5
Date: Fri, 26 Mar 2010 11:57:49 +0530
From: Anil Gupta <anilgupta83 at gmail.com>
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: asterisk-ss7 at lists.digium.com
Message-ID:
    <3bb51fcb1003252327h585a7ed9r60d0627a0166aef1 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

oops seems you already did the recording thing.

Do you mean only one way audio path(from isdn to ss7) is open before ANM ?


On Fri, Mar 26, 2010 at 11:09 AM, Anil Gupta <anilgupta83 at gmail.com> wrote:

> Hi,
>
> I doubt ss7 starts transmitting audio because I can see asterisk does not
> actually include optional backward call inband information parameter,
inband
> information parameter should be set to 1 for audio path to open before
ANM.
> I believe some switches could be configured to force this behavior. You
> might want to record on a dahdi channel to make sure if audio is being
> transmitted after ACM
>
> Regards,
>
> Anil
>
> PS : Its will be better if you start a new thread for this.
>
> On Fri, Mar 26, 2010 at 4:17 AM, Domjan Attila <adomjan at tvnet.hu> wrote:
>
>> Hi,
>> I'm not sure it should work... it is not ss7 it is dahdi issue....
>>
>> On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:
>> > Hi Attila, your patch more one time work without problems thank you
>> again.
>> >
>> > Regarding this email let me check other problem with you. I have a
>> asterisk
>> > box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN
>> to
>> > CISCO
>> >
>> > TDM<----SS7--->ASTERISK<---ISDN---> CISCO
>> >
>> > The call come from SS7 side and asterisk forward this call to ISDN. The
>> > cisco gateway plays a message in PROCEEDING. This message ask the user
>> to
>> > put some digits. The calling side listen the message but when he put
the
>> > digits the ISDN side don't receive the DTMF.
>> >
>> > I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I
>> can
>> > listen the audio and the dtmf and in ISDN side I don't listen the DTMF
>> only
>> > the audio
>> >
>> > I don't know why asterisk don't foward the audio received on SS7 side
>> before
>> > the ANM  during the early media.
>> >
>> > Do you know anything about this ?
>> >
>> >
>> > Regards,
>> > Bruno Rodrigues
>> >
>> > --------------------------------------------------
>> > From: "Attila Domjan" <adomjan at tvnet.hu>
>> > Sent: Wednesday, March 17, 2010 9:17 AM
>> > To: <asterisk-ss7 at lists.digium.com>
>> > Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
>> >
>> > > --
>> > > _____________________________________________________________________
>> > > -- 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...
URL:
http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/1374a1a1
/attachment.htm 

------------------------------

_______________________________________________
--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

End of asterisk-ss7 Digest, Vol 61, Issue 34
********************************************



      
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/343fcb0e
/attachment.htm 

------------------------------

_______________________________________________
--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

End of asterisk-ss7 Digest, Vol 61, Issue 35
********************************************

More information about the asterisk-ss7 mailing list