Re: [asterisk-users] Connecting two Asterisk
Some other data... I changed both iax.conf and wrote: bandwidth=high allow=all Now I see in the log: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (), priority = mine and I can hear somewhat, but with a VERY poor quality on my mobile phone... On the other phone however, the quality is very good... I'm very very puzzled... Thanks for any help! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two Asterisk
Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME configuration!!) installed on my Server... Well, I will try to configure the Asterisk on my Server to act as proxy so that all phones at home talk with my Asterisk at home (now called wrt, and my mobile phone talk with my Asterisk on my server (now called lucabert). I followed this HowTo http://sysmagazine.com/posts/125303/ and I got both Servers talking together. I can call my mobile phone (logged in at lucabert) from a phone logged in on wrt and a phone at wrt from my mobile phone at lucabert. Wonderful! Now the problem: on my phones at wrt I can hear what the mobile phone at lucabert sends (with a very good audio-quality), but on this mobile phone I cannot hear a single word spoken with the phone at wrt, not even the music on hold I configured... When I call my mobile phone from a phone logged on at wrt I see on the Asterisk at wrt: == Using SIP RTP CoS mark 5 -- Executing [4@default:1] Verbose(SIP/004935-001e, 2,Internal call for Mobile - [004935]) in new stack == Internal call for Mobile - [004935] -- Executing [4@default:2] Dial(SIP/004935-001e, IAX2/lucabert:MYVERYSECRET@lucabert/0049177333,,R) in new stack -- Called IAX2/lucabert:MYVERYSECRET@lucabert/0049177333 -- Call accepted by X.Y.Z.K (format gsm) -- Format for call is gsm -- IAX2/lucabert-1298 is ringing -- IAX2/lucabert-1298 answered SIP/004935-001e -- Started music on hold, class 'default', on IAX2/lucabert-1298 -- Stopped music on hold on IAX2/lucabert-1298 -- Hungup 'IAX2/lucabert-1298' == Spawn extension (default, 4, 2) exited non-zero on 'SIP/004935-001e' On the Asterisk at lucabert I see: -- Accepting AUTHENTICATED call from A.B.C.D: requested format = ulaw, requested prefs = (ulaw|gsm|g729|alaw), actual format = gsm, host prefs = (gsm|g729|alaw|ulaw), priority = mine -- Executing [0049177333@default:1] Macro(IAX2/lucabert-94, stdexten,0049177333,SIP/0049177333DAHDI/1) in new stack [Jun 7 21:59:09] WARNING[19888]: app_macro.c:302 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten' -- Executing [0049177333@default:2] Set(IAX2/lucabert-94, CHANNEL(musicclass)=default) in new stack -- Executing [0049177333@default:3] Dial(IAX2/lucabert-94, SIP/0049177333,,R) in new stack == Using SIP RTP CoS mark 5 -- Called 0049177333 -- SIP/0049177333-0008 is ringing -- SIP/0049177333-0008 is ringing -- SIP/0049177333-0008 is ringing -- SIP/0049177333-0008 answered IAX2/lucabert-94 == Spawn extension (default, 0049177333, 3) exited non-zero on 'IAX2/lucabert-94' -- Hungup 'IAX2/lucabert-94' Well, I'm very puzzled... Can someone help me? Thank you very much! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk
Steve Edwards asterisk@sedwards.com schrieb: On Sun, 7 Jun 2015, Luca Bertoncello wrote: Now the problem: on my phones at wrt I can hear what the mobile phone at lucabert sends (with a very good audio-quality), but on this mobile phone I cannot hear a single word spoken with the phone at wrt, not even the music on hold I configured... -- Call accepted by X.Y.Z.K (format gsm) -- Format for call is gsm I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw? Yes, I do, but the quality is always very poor... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk
On Sun, 7 Jun 2015, Luca Bertoncello wrote: I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME configuration!!) installed on my Server... Maybe fiddling with the SIP and RTP ports would help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk
On Sun, 7 Jun 2015, Luca Bertoncello wrote: Now the problem: on my phones at wrt I can hear what the mobile phone at lucabert sends (with a very good audio-quality), but on this mobile phone I cannot hear a single word spoken with the phone at wrt, not even the music on hold I configured... -- Call accepted by X.Y.Z.K (format gsm) -- Format for call is gsm I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two Asterisk together via SIP + DISA
Hi all, I need to test the following scenario: +---+ +---+ | asterisk 1| | asterisk 2| +---+ +---+ | | | | ___|__|___ | | | | | | +---+ +---+ | ATA 1 | | ATA 2 | +---+ +---+ / \ / \ /\ /\ 21 22 1011 That is, I have 2 asterisks connected via SIP, two ATAs with two lines, and the ATA1 is registered with asterisk1 and ATA2 is registered with asterisk2, and all incoming calls in asterisk2 from the asterisk1 (via SIP), are answered by a DISA. I can make calls between ATA1 and ATA2 without problems (the call will be routed to the asterisk1 to asterisk2, falls in DISA and I call one of the phones ATA2). I am now trying to make the call coming from,eg, extension 21, go to the asterisk1 - asterisk2, answered by the DISA and go back asterisk1, ringing the branch 22. Since I am newbie in this matter, I wonder with friends from the list if this is possible ... Or is there another way to do this Below is my conf files. Rgs Cesar === asterisk 1 ** sip.conf [21] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=21 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen; Set caller ID presentation ; See doc/callingpres.txt for more information [22] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=22 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default
[asterisk-users] Connecting two Asterisk together via SIP + DISA
Hi all, I need to test the following scenario: +---+ +---+ | asterisk 1| | asterisk 2| +---+ +---+ | | | | ___|__|___ | | | | | | +---+ +---+ | ATA 1 | | ATA 2 | +---+ +---+ / \ / \ /\ /\ 21 22 1011 That is, I have 2 asterisks connected via SIP, two ATAs with two lines, and the ATA1 is registered with asterisk1 and ATA2 is registered with asterisk2, and all incoming calls in asterisk2 from the asterisk1 (via SIP), are answered by a DISA. I can make calls between ATA1 and ATA2 without problems (the call will be routed to the asterisk1 to asterisk2, falls in DISA and I call one of the phones ATA2). I am now trying to make the call coming from,eg, extension 21, go to the asterisk1 - asterisk2, answered by the DISA and go back asterisk1, ringing the branch 22. Since I am newbie in this matter, I wonder with friends from the list if this is possible ... Or is there another way to do this Below is my conf files. Rgs Cesar === asterisk 1 ** sip.conf [21] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=21 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen; Set caller ID presentation ; See doc/callingpres.txt for more information [22] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=22 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection
MOSBAH ABDELKADER wrote: Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. I never do that. I use a router that supports Frame Relay. For me, installing a Digium card just to connect to a Frame Relay network is much more work, poorly documented, and just much more hassle than using a router with Frame Relay support. You could not use any of the TDM cards, you would need a T-1/E-1 card from Digium or Sangoma if you wanted to go that route. I would not recommend it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay
Hello, As we know, to connect Asterisk to PSTN network, we must use a PCI card containing FXS and FXO modules like Digium TDM400P. Now to connect Asterisk to a Frame Relay network what is the PCI card that we need? Is the Ethernet adapter only is enough? or i have to buy another type of PCI card?. Thanks. Mosbah Abdelkader. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection
easiest way of connecting multiple Asterisk boxes are trough IP network. I know Digium cards supports HDLC encapsulation but i'm not sure about framerelay. On 8/4/07, Michael Munger [EMAIL PROTECTED] wrote: What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *MOSBAH ABDELKADER *Sent:* Saturday, August 04, 2007 3:16 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two Asterisk servers with a frame relay connection
Hello all, I have to connect two Asterisk servers with a frame relay connection but i do not know what is the hardware to use and how to connect them. Have anyone an idea about that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection
You could use SIP if the servers are on routable IPs or the same subnet, if not you could use IAX but I think OpenVPN is your best choice for using SIP over different NATed networks. I do not think you need any hardware except for what is needed for the Frame Relay. QoS and traffic shaping would be a good idea if other traffic is going over your link. Thanks, Steve Totaro MOSBAH ABDELKADER wrote: Hello all, I have to connect two Asterisk servers with a frame relay connection but i do not know what is the hardware to use and how to connect them. Have anyone an idea about that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection
Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection
What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two asterisk server.
Sir, I want to conact two asterisk server to communicate each other and i want to make all outgoing call perform only one server for example i have two server A and B i have attach TDM02B card on server A. first i want to connect both server. second i want to make all outgoing call perform by server A suppose if the user of server B want to call outside than call perfrom from server A because server B having no PSTN line all PSTN line are on server A. is it possible if yes please explain step by step i am working on SIP protocol. server A and server B running fine without connecting each other. Rajeev. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two asterisk servers
Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two asterisk servers
Please search the wiki first. Most of your questions you post can easily be found by doing a search. Put some effort into finding the answers to your questions first and on your own, and then if you still have questions, I'm sure everyone would be more than willing to help. On 8/29/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you. With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Connecting two asterisk servers
In short, yes... The wiki (http://www.voip-info.org) has documentation on how to configure your servers, how to configure the dialplan, etcI don't mean to single you out mate, but has anyone else noticed an increase in the number of questions being asked that could have been answered simply by visiting the wiki, reading the sample docs in the package, or even doing a Google search? I seem to recall the general rule of this list is that you should have already at least tried to find the answer. Here's a few links to get you started: The Asterisk Wiki, Asterisk Guru, Getting Started, GNU Inter, AGI Guide, O'reilly Onlamp Article - by John Todd, One Unified. It took me more time to cut and past those links than it did to find them, they were on the Asterisk.org support page. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Tuesday, August 29, 2006 11:16 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Connecting two asterisk servers Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two asterisk servers
Crazy Boy a écrit : Hi friends, Thank you to all for your response and cooperation to me. I have a doubt. I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers? Yes. Just have something like: [serverA] type=peer host=serverA.IP.Address In ServerB's sip.conf and [serverB] type=peer host=serverB.IP.Address In ServerA's sip.conf 2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)? Of course. Say user joe is registered with serverB, then within serverA's dialplan, you can use: exten = 123456,1,Dial(SIP/[EMAIL PROTECTED]) ; [EMAIL PROTECTED] has extension '123456' Within serverB's dialplan, you'd simply use: exten = 123456,1,Dial(SIP/joe) ; [EMAIL PROTECTED] has extension '123456' Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting two asterisk servers using IAX
I am trying to connect two asterisk servers using the information from: http://www.voip-info.org/tiki-index.php ... +2+servers It works fine with Method 1. If I use method 3, I get errors: on sending server: Registration of 'REC_SERVER' rejected: Registration Refused on receiving server: No registration for peer 'REC_SERVER' (from x.x.x.x) Any thoughts? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
