<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:arial,helvetica,sans-serif;font-size:12pt"><div><br></div><div style="font-family: arial,helvetica,sans-serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> "asterisk-ss7-request@lists.digium.com" <asterisk-ss7-request@lists.digium.com><br><b><span style="font-weight: bold;">To:</span></b> asterisk-ss7@lists.digium.com<br><b><span style="font-weight: bold;">Sent:</span></b> Fri, March 26, 2010 8:27:57 AM<br><b><span style="font-weight: bold;">Subject:</span></b> asterisk-ss7 Digest, Vol 61, Issue 34<br></font><br>
Send asterisk-ss7 mailing list submissions to<br> <a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br><span> <a target="_blank" href="http://lists.digium.com/mailman/listinfo/asterisk-ss7">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></span><br>or, via email, send a message with subject or body 'help' to<br> <a ymailto="mailto:asterisk-ss7-request@lists.digium.com" href="mailto:asterisk-ss7-request@lists.digium.com">asterisk-ss7-request@lists.digium.com</a><br><br>You can reach the person managing the list at<br> <a ymailto="mailto:asterisk-ss7-owner@lists.digium.com" href="mailto:asterisk-ss7-owner@lists.digium.com">asterisk-ss7-owner@lists.digium.com</a><br><br>When replying, please edit your Subject line so it is more
specific<br>than "Re: Contents of asterisk-ss7 digest..."<br><br><br>Today's Topics:<br><br> 1. Re: IAM-REL instead of IAM-ACM-REL (Bruno Rodrigues de Mello)<br> 2. Re: IAM-REL instead of IAM-ACM-REL (Domjan Attila)<br> 3. Where's Matt been? Well, here's the explanation<br> (Matthew Fredrickson)<br> 4. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)<br> 5. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Thu, 25 Mar 2010 15:07:30 -0300<br>From: "Bruno Rodrigues de Mello" <<a ymailto="mailto:shotsbros@hotmail.com" href="mailto:shotsbros@hotmail.com">shotsbros@hotmail.com</a>><br>Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>To: <<a ymailto="mailto:asterisk-ss7@lists.digium.com"
href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a>><br>Message-ID: <<a ymailto="mailto:SNT127-DS14B34BD7A59DD4759F3770B0240@phx.gbl" href="mailto:SNT127-DS14B34BD7A59DD4759F3770B0240@phx.gbl">SNT127-DS14B34BD7A59DD4759F3770B0240@phx.gbl</a>><br>Content-Type: text/plain; format=flowed; charset="utf-8";<br> reply-type=original<br><br>Hi Attila, your patch more one time work without problems thank you again.<br><br>Regarding this email let me check other problem with you. I have a asterisk <br>box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN to <br>CISCO<br><br>TDM<----SS7--->ASTERISK<---ISDN---> CISCO<br><br>The call come from SS7 side and asterisk forward this call to ISDN. The <br>cisco gateway plays a message in PROCEEDING. This message ask the user to <br>put some digits. The calling side listen the message but when he put the <br>digits the ISDN side don't
receive the DTMF.<br><br>I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I can <br>listen the audio and the dtmf and in ISDN side I don't listen the DTMF only <br>the audio<br><br>I don't know why asterisk don't foward the audio received on SS7 side before <br>the ANM during the early media.<br><br>Do you know anything about this ?<br><br><br>Regards,<br>Bruno Rodrigues<br><br>--------------------------------------------------<br>From: "Attila Domjan" <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>><br>Sent: Wednesday, March 17, 2010 9:17 AM<br>To: <<a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a>><br>Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br><br>> -- <br>> _____________________________________________________________________<br><span>> -- Bandwidth and Colocation
Provided by <a target="_blank" href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br>><br>> asterisk-ss7 mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a> <br><br><br><br><br>------------------------------<br><br>Message: 2<br>Date: Thu, 25 Mar 2010 23:47:30 +0100<br>From: Domjan Attila <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>><br>Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>To: <a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a><br>Message-ID: <1269557250.2046.1.camel@localhost><br>Content-Type: text/plain; charset="utf-8"<br><br>Hi,<br>I'm not sure it should work... it is not ss7 it is dahdi
issue....<br><br>On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:<br>> Hi Attila, your patch more one time work without problems thank you again.<br>> <br>> Regarding this email let me check other problem with you. I have a asterisk <br>> box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN to <br>> CISCO<br>> <br>> TDM<----SS7--->ASTERISK<---ISDN---> CISCO<br>> <br>> The call come from SS7 side and asterisk forward this call to ISDN. The <br>> cisco gateway plays a message in PROCEEDING. This message ask the user to <br>> put some digits. The calling side listen the message but when he put the <br>> digits the ISDN side don't receive the DTMF.<br>> <br>> I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I can <br>> listen the audio and the dtmf and in ISDN side I don't listen the DTMF only <br>> the audio<br>> <br>> I don't know
why asterisk don't foward the audio received on SS7 side before <br>> the ANM during the early media.<br>> <br>> Do you know anything about this ?<br>> <br>> <br>> Regards,<br>> Bruno Rodrigues<br>> <br>> --------------------------------------------------<br>> From: "Attila Domjan" <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>><br>> Sent: Wednesday, March 17, 2010 9:17 AM<br>> To: <<a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a>><br>> Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>> <br>> > -- <br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>> ><br>> > asterisk-ss7
mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a> <br>> <br>> <br><br>-------------- next part --------------<br>A non-text attachment was scrubbed...<br>Name: not available<br>Type: application/pgp-signature<br>Size: 198 bytes<br>Desc: This is a digitally signed message part<br><span>Url : <a target="_blank" href="http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100325/30e7c1e7/attachment-0001.pgp">http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100325/30e7c1e7/attachment-0001.pgp</a> </span><br><br>------------------------------<br><br>Message: 3<br>Date: Thu, 25 Mar 2010 23:40:29 -0500<br>From: Matthew Fredrickson <<a ymailto="mailto:creslin@digium.com" href="mailto:creslin@digium.com">creslin@digium.com</a>><br>Subject: [asterisk-ss7] Where's
Matt been? Well, here's the<br> explanation<br>To: <a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a><br>Message-ID: <<a ymailto="mailto:4BAC3ABD.4000701@digium.com" href="mailto:4BAC3ABD.4000701@digium.com">4BAC3ABD.4000701@digium.com</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Hey all,<br><br>It's been a long time. I apologize for my quietness for the last while <br>here, it has been a very busy year this last year with some of the other <br>projects I've been working on.<br><br>I just wanted to let everyone know that I'm still alive, and haven't <br>given up or forgotten about libss7, and Asterisk with SS7.<br><br>I appreciate very much Attila's great work on getting libss7 polished <br>up. He's done a very good job taking it to the next level, and has done <br>a great job helping everyone out.
Thanks very much Attila.<br><br>As far as getting his code merged back in, I was in the midst of this a <br>while ago, but had to stop due to personal time issues as well as some <br>other potentially architectural issues that subsequently came up. <br>Hopefully we can get that moving again in the next little bit though.<br><br>I actually have been quite busy, and hopefully have some things to show <br>for it, two things actually.<br><br>I'll not go into too much detail tonight (as it's getting quite late) <br>but I'm looking for some hungry testers, that wouldn't mind beating up <br>on some probably alpha level code.<br><br>They are:<br><br>1.) libss7 point code clustering support.<br><br>Basically, you can have Asterisk boxes share signalling links now using <br>this code. Although the signalling links are physically terminated on <br>other machines, you can plug bearer T1s/E1s into other Asterisk boxes <br>and virtually utilize the signalling
links of the other boxes.<br><br>2.) A new channel driver, called chan_ccs, that allows, among other <br>things, you to control MGCP media gateways for bearer trunks, instead of <br>having to locally terminate them on the asterisk box that's controlling <br>the signalling links. There is also code in the same branch that has <br>chan_ccs that modified chan_mgcp so that Asterisk can act as a media <br>gateway (since I don't have any good real media gateways to test on). <br>This basically means you can have Asterisk TDM channel scalability up to <br>(in the ideal state) the same level as you can do with SIP with no <br>media, per box.<br><br>In essence, this is turning Asterisk into a "true" softswitch, allowing <br>native bridging between media gateways and any other RTP endpoint <br>(including other media gateways). This also means that you don't have <br>to terminate bearer T1s/E1s on the main signalling box.<br><br>-- So, what does this
mean for you, you may be asking?<br><br>These are really the next steps in making big SS7 work with Asterisk. <br>They both allow for scaling a point code across multiple asterisk <br>machines, and distribution of bearers on different boxes than the ones <br>that contain signalling links.<br><br>Like I said though, most of the work is in a functional but early phase, <br>and so I need some people that are interested enough in the added <br>functionality that they're willing to work with potential hickups along <br>the way.<br><br>Some of the changes I've had to make to libss7 have made it further more <br>difficult to merge Attila's changes back in, which is the other reason <br>why it has been so long and it still has not been merged.<br><br>If you're interested, either reply to me or this thread and let me know.<br><br>Thanks again,<br><br>Matthew Fredrickson<br>Digium, Inc.<br><br><br><br><br>------------------------------<br><br>Message: 4<br>Date:
Fri, 26 Mar 2010 11:09:48 +0530<br>From: Anil Gupta <<a ymailto="mailto:anilgupta83@gmail.com" href="mailto:anilgupta83@gmail.com">anilgupta83@gmail.com</a>><br>Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>To: <a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a><br>Message-ID:<br> <<a ymailto="mailto:3bb51fcb1003252239s6542a998w21f6105261190a5f@mail.gmail.com" href="mailto:3bb51fcb1003252239s6542a998w21f6105261190a5f@mail.gmail.com">3bb51fcb1003252239s6542a998w21f6105261190a5f@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hi,<br><br>I doubt ss7 starts transmitting audio because I can see asterisk does not<br>actually include optional backward call inband information parameter, inband<br>information parameter should be set to 1 for audio path to open before ANM.<br>I believe some switches could be
configured to force this behavior. You<br>might want to record on a dahdi channel to make sure if audio is being<br>transmitted after ACM<br><br>Regards,<br><br>Anil<br><br>PS : Its will be better if you start a new thread for this.<br><br>On Fri, Mar 26, 2010 at 4:17 AM, Domjan Attila <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>> wrote:<br><br>> Hi,<br>> I'm not sure it should work... it is not ss7 it is dahdi issue....<br>><br>> On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:<br>> > Hi Attila, your patch more one time work without problems thank you<br>> again.<br>> ><br>> > Regarding this email let me check other problem with you. I have a<br>> asterisk<br>> > box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN<br>> to<br>> > CISCO<br>> ><br>> > TDM<----SS7--->ASTERISK<---ISDN--->
CISCO<br>> ><br>> > The call come from SS7 side and asterisk forward this call to ISDN. The<br>> > cisco gateway plays a message in PROCEEDING. This message ask the user to<br>> > put some digits. The calling side listen the message but when he put the<br>> > digits the ISDN side don't receive the DTMF.<br>> ><br>> > I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I<br>> can<br>> > listen the audio and the dtmf and in ISDN side I don't listen the DTMF<br>> only<br>> > the audio<br>> ><br>> > I don't know why asterisk don't foward the audio received on SS7 side<br>> before<br>> > the ANM during the early media.<br>> ><br>> > Do you know anything about this ?<br>> ><br>> ><br>> > Regards,<br>> > Bruno Rodrigues<br>> ><br>> > --------------------------------------------------<br>> > From: "Attila
Domjan" <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>><br>> > Sent: Wednesday, March 17, 2010 9:17 AM<br>> > To: <<a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a>><br>> > Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>> ><br>> > > --<br>> > > _____________________________________________________________________<br>> > > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>> > ><br>> > > asterisk-ss7 mailing list<br>> > > To UNSUBSCRIBE or update options visit:<br>> > > <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>> ><br>>
><br>><br>><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>><br>> asterisk-ss7 mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br><span>URL: <a target="_blank" href="http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/5ad6756d/attachment-0001.htm">http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/5ad6756d/attachment-0001.htm</a> </span><br><br>------------------------------<br><br>Message: 5<br>Date: Fri, 26 Mar 2010 11:57:49 +0530<br>From: Anil Gupta <<a
ymailto="mailto:anilgupta83@gmail.com" href="mailto:anilgupta83@gmail.com">anilgupta83@gmail.com</a>><br>Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>To: <a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a><br>Message-ID:<br> <<a ymailto="mailto:3bb51fcb1003252327h585a7ed9r60d0627a0166aef1@mail.gmail.com" href="mailto:3bb51fcb1003252327h585a7ed9r60d0627a0166aef1@mail.gmail.com">3bb51fcb1003252327h585a7ed9r60d0627a0166aef1@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>oops seems you already did the recording thing.<br><br>Do you mean only one way audio path(from isdn to ss7) is open before ANM ?<br><br><br>On Fri, Mar 26, 2010 at 11:09 AM, Anil Gupta <<a ymailto="mailto:anilgupta83@gmail.com" href="mailto:anilgupta83@gmail.com">anilgupta83@gmail.com</a>> wrote:<br><br>> Hi,<br>><br>> I doubt
ss7 starts transmitting audio because I can see asterisk does not<br>> actually include optional backward call inband information parameter, inband<br>> information parameter should be set to 1 for audio path to open before ANM.<br>> I believe some switches could be configured to force this behavior. You<br>> might want to record on a dahdi channel to make sure if audio is being<br>> transmitted after ACM<br>><br>> Regards,<br>><br>> Anil<br>><br>> PS : Its will be better if you start a new thread for this.<br>><br>> On Fri, Mar 26, 2010 at 4:17 AM, Domjan Attila <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>> wrote:<br>><br>>> Hi,<br>>> I'm not sure it should work... it is not ss7 it is dahdi issue....<br>>><br>>> On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:<br>>> > Hi Attila, your patch more one time work
without problems thank you<br>>> again.<br>>> ><br>>> > Regarding this email let me check other problem with you. I have a<br>>> asterisk<br>>> > box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN<br>>> to<br>>> > CISCO<br>>> ><br>>> > TDM<----SS7--->ASTERISK<---ISDN---> CISCO<br>>> ><br>>> > The call come from SS7 side and asterisk forward this call to ISDN. The<br>>> > cisco gateway plays a message in PROCEEDING. This message ask the user<br>>> to<br>>> > put some digits. The calling side listen the message but when he put the<br>>> > digits the ISDN side don't receive the DTMF.<br>>> ><br>>> > I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I<br>>> can<br>>> > listen the audio and the dtmf and in ISDN side I don't listen the DTMF<br>>>
only<br>>> > the audio<br>>> ><br>>> > I don't know why asterisk don't foward the audio received on SS7 side<br>>> before<br>>> > the ANM during the early media.<br>>> ><br>>> > Do you know anything about this ?<br>>> ><br>>> ><br>>> > Regards,<br>>> > Bruno Rodrigues<br>>> ><br>>> > --------------------------------------------------<br>>> > From: "Attila Domjan" <<a ymailto="mailto:adomjan@tvnet.hu" href="mailto:adomjan@tvnet.hu">adomjan@tvnet.hu</a>><br>>> > Sent: Wednesday, March 17, 2010 9:17 AM<br>>> > To: <<a ymailto="mailto:asterisk-ss7@lists.digium.com" href="mailto:asterisk-ss7@lists.digium.com">asterisk-ss7@lists.digium.com</a>><br>>> > Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL<br>>> ><br>>> > > --<br>>> > >
_____________________________________________________________________<br>>> > > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>>> > ><br>>> > > asterisk-ss7 mailing list<br>>> > > To UNSUBSCRIBE or update options visit:<br>>> > > <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>>> ><br>>> ><br>>><br>>><br>>> --<br>>><br>>> _____________________________________________________________________<br>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>>><br>>> asterisk-ss7 mailing list<br>>> To UNSUBSCRIBE or update options visit:<br>>> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>>><br>><br>><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br><span>URL: <a target="_blank" href="http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/1374a1a1/attachment.htm">http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/1374a1a1/attachment.htm</a> </span><br><br>------------------------------<br><br>_______________________________________________<br><span>--Bandwidth and Colocation Provided by <a target="_blank" href="http://www.api-digital.com">http://www.api-digital.com</a>--</span><br><br>asterisk-ss7 mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br><br>End of asterisk-ss7
Digest, Vol 61, Issue 34<br>********************************************<br></div></div>
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