Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk.
I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak ----- Original Message ----- From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, January 20, 2004 12:08 PM Subject: Asterisk-Users digest, Vol 1 #2557 - 10 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. FW: Memory problem (T. Chan) > 2. X101P CallOut Big Problem. (Carlos Arnt) > 3. RE: RE: Latest version of asterisk (Aram Ter-Martirosyan) > 4. Re: SIP: Register that isn't a register? (Ing. Angel Gomez Garcia) > 5. RE: FW: Memory problem (Adam Goryachev) > 6. Call token is ip$localhost (Asan M.) > 7. Re: CVS Changes (NAT-SIP) (Brian West) > 8. Re: PLAYBACK multiple files (Marcin Kuzmicki) > 9. Re: user password and call waiting (Brian West) > 10. echo cancellation (dkwok) > > --__--__-- > > Message: 1 > From: "T. Chan" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Mon, 19 Jan 2004 23:20:27 -0500 > Subject: [Asterisk-Users] FW: Memory problem > Reply-To: [EMAIL PROTECTED] > > > Dear all, > > I have had an experience which I would run by all of you to see if this is > normal. > > I am running a few asterisk servers with 512M RAM memory, and as I have > mentioned in previous notes, I have experienced frequent crashes when faced > with more than 15-20 simultaneous calls. I have tried to find out if it > could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3, > (c) old redhat linux version 7.3, (d) H323 library pwlib and openh323 > versions which are 1.5.2 and 1.12.2 respectively among many other > parameters. So far, unfortunately, the matter has not been resolved. > However, I have noticed that the memory usage on each server has built up > with time after the server being rebooted. I have complained about using > close to 500M even when there were very few calls on the server but nobody > seemed to be able to let me know if they were running at high memory usages > except for Jesse who was telling me that his memory usages have always been > low. Very recently, I noticed that after I rebooted the servers, the memory > usage would start at about 80 M and even after started the Asterisk threads, > I was running at about 100 M and even when there were calls, I was running > at about 100M-150M, but then after hours it would start to build up to 200M > and then 250M and then....finally close to 500M even after I stopped the > Asterisk threads, almost like there is a memory leak somewhere. > > I wonder if that is normal, if someone can please tell me, or if not normal, > what could be the cause to it and how should this be rectified. > > Thanks alot > > > Tom > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 > > > --__--__-- > > Message: 2 > From: Carlos Arnt <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Tue, 20 Jan 2004 02:39:09 -0200 > Subject: [Asterisk-Users] X101P CallOut Big Problem. > Reply-To: [EMAIL PROTECTED] > > <html><head><meta name=3D"Generator" content=3D"PocoMail 3 HTML/CSS= > Generator"/> > <style type=3D"text/css"><!-- > LI{display:list-item;margin:0.00in;} > p{display:block;margin:0.00in;} > body{} > --></style> > </head><BODY ><p><SPAN style=3D"font-size:10pt;">Hi all,</SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">I just now receive the FXO X101P Card but= > can't at any way make then call out.</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">I can hear the signal, even call but always= > receive from my local operator error that or the number don't exist or need= > more numbers.</SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">I play alot with txgain and rxgain, but= > none help me out.</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">Being honest i try alot !!!! 5 hours and= > none !!!</SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">I'm using asterisk in his sample= > configs.</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">I mean i call out using 1234= > etc..</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">Zapata.conf is Ok</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">Zaptel.conf is ok</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">(I follow the Digium faqs, then for a good= > person that show-me this in the Asterisk IRC)</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">( Using here is an Asterisk= > 7.1)</SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">Did anyone know a txgain and rxgain from= > Brazilian lines ? (I'm trying with Vesper operator)</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">Did i need make something more ( i know= > that need) :)</SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">Please could someone with lot's of time= > help-me out here with this simple question ?</SPAN></p> > <p><SPAN style=3D"font-size:10pt;">I just wanna call out too !!! </SPAN></p> > <p> </p> > <p><SPAN style=3D"font-size:10pt;">Thanks alot !</SPAN></p> > <p> </p> > </body></html> > > > --__--__-- > > Message: 3 > From: "Aram Ter-Martirosyan" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] RE: Latest version of asterisk > Date: Mon, 19 Jan 2004 20:42:26 -0800 > Reply-To: [EMAIL PROTECTED] > > Hello Matt, > Is that the Wildcard TE410P you are using. Digium said that it had some > problems with Redhat 9.0 is that correct? > > - Digium quad T1 card > - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) > - Redhat 9.0 > > Aram Ter-Martirosyan > Senior Account Manager > Hi-Tech Gateway, Inc. > http://www.hi-teck.com > 1225 Grand Central Ave. > Glendale, CA 91201 > [EMAIL PROTECTED] > tel 818.546.4601 > fax 818.546.4617 > Turning Technology Into Business Solutions > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of mattf > Sent: Monday, January 19, 2004 6:21 PM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] RE: Latest version of asterisk > > > Hello, > > Our max for a single machine is 40 concurrent SIP -> Zap conversations for > about a 12 hour period and over 5000 total phone calls per day. We didn't > see crashes going over that, but we wanted to be safe and now have 2 > identical machines handling upto about 30 concurrent SIP -> Zap calls(3000 > phone calls per day), and a third old machine for office use that never gets > over 10 concurrent calls. Here's the specs for these systems: > > - 120 installed hardphones: > - 80 x grandstream 102 hardphones > - 20 x Sipura analog adapters(2 phones each) > - 2 x Asterisk servers > - 2.6 GHz Pentium4 800MHz bus w/ HyperThreading enabled > - Asus p4c800 800MHz mobo > - 2GB DDR400 RAM (This is actually overkill you need 1GB max if you > reboot weekly) > - 4 x 36GB SCSI drives in RAID 10 w/megaraid card > - 3com 905CX ethernet card > - Digium quad T1 card > - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) > - Redhat 9.0 > - Asterisk with many modules turned off and no MOH > > With these servers you can see the load average jump from 0.00 to 6.25 in a > matter of a minute and then back down again, all while never dropping a call > or crashing. > > We also recently diagnosed our lock-freeze to the touchy manager > interface(if you are logged into the manager interface and you loose > connection, the manager outgoing buffer seems to overflow and freeze > Asterisk). So it doesn't seem to be a problem of hardware. But we still > haven't figured out how to fix it. > > One note as to Ethernet cards, we actually fried a Realtek 8139 Ethernet > card that we had put in a server temporarily as we were doing our testing. > It started to generate a lot of errors and dropping packets left and right. > When we took it out it was VERY hot. We then put in a 3com 905 card and > haven't had an issue with it yet. > > Hope this helps, > > MATT--- > > > > -----Original Message----- > From: T. Chan [mailto:[EMAIL PROTECTED] > Sent: Monday, January 19, 2004 4:49 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] RE: Latest version of asterisk > > > Thanks, Matt ! > > So, am I correct in assuming that there are quite a few (or alot) of us who > have had not so good experiences with Asterisk? That Asterisk would crash > after it hit a certain number of calls or after a certain period of time > with 15-20 calls? I understand that there were others who were able to send > a good number of calls through but can anyone tell us if they have had > tested and confirmed that Asterisk runs better without or with HT and in > terms of number of calls, how many would each one support, in the ballpark? > It would also be nice if one could tell us the computer configuration in > order to send that many calls without crashing Asterisk. Does it make a > difference running the LAN on a ONBOARD LAN card as compared to a PCI Intel > or 3COM LAN card, since there is a chance that packets are passing more > efficiently on a PCI LAN card? > > Side question: Is it possible to do passthrough faxing? Like, customers > sending me H323 or SIP fax calls and the Asterisk will pass through to > another gateway? Anyone successful in doing that? > > Tommy > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of mattf > Sent: Monday, January 19, 2004 8:32 AM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] RE: Latest version of asterisk > > > Hello, > > I've had Asterisk installed on HT capable machines in both HT mode(with SMP) > and non HT mode (with non-SMP) and did not notice any differences > functionally between them. The processor load was always less in HT SMP mode > than non HT and I have experienced Asterisk deadlocks in both modes so it > doesn't really seem to matter if you leave HT on(at least in my > experiences). > > HT basically works by splitting off commands to one of two different virtual > processors that both run at about 70% of processor's speed(that's why you > may notice compiling to take longer when in HT mode) I have heard of some > applications having memory addressing errors with HT but I have not seen any > evidence to support that in Asterisk thus far. > > I'm going to try installing a 4 x T1 card on my Athlon 2xMP server next week > and see if Asterisk/Digium performance/compatibility improves over the Intel > platform. > > > MATT--- > > > -----Original Message----- > From: WipeOut [mailto:[EMAIL PROTECTED] > Sent: Monday, January 19, 2004 2:54 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] RE: Latest version of asterisk > > > T. Chan wrote: > > >Dear All > > > >Should one enable HT in the chip when running Asterisk or if we don't, > would > >that offer alot less processing power? > > > >T > > > I have read before that HT did not help Asterisk so should be dissabled, > but as the chipsets and other hardware get better at using and > controlling HT it may help.. > > Run some tests on your system and see what your conclusions are, then > feedback your findings to the list so that others may learn from it.. > > Later.. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > Date: Mon, 19 Jan 2004 21:22:39 -0800 > From: "Ing. Angel Gomez Garcia" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] SIP: Register that isn't a register? > To: [EMAIL PROTECTED] > Reply-To: [EMAIL PROTECTED] > > Walter Doerr wrote: > > >On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote: > > > > > >>Ok, > >> > >>here comes part two of the log quiz, this time SIP not MGCP: > >> > >>WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER > >>that isn't a register > >> > >>This is most probably cause by registration of * with FWD. > >> > >> > > > >I am seeing this with iptel.org > > > >-Walter > > > > > I had this when registering to FWD from * inside my LAN and without > externip configured, If * sends its internal IP, the FWD server returns > this message. > > > --__--__-- > > Message: 5 > From: "Adam Goryachev" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] FW: Memory problem > Date: Tue, 20 Jan 2004 16:16:14 +1100 > Reply-To: [EMAIL PROTECTED] > > > > [EMAIL PROTECTED] <> wrote: > > I am running a few asterisk servers with 512M RAM memory, and > > as I have > > mentioned in previous notes, I have experienced frequent > > crashes when faced > > with more than 15-20 simultaneous calls. I have tried to find > > out if it > > could be due to (a) Xeon chip running HT, (b) old Kernel > > version 2.4.18-3, > > (c) old redhat linux version 7.3, (d) H323 library pwlib and openh323 > > versions which are 1.5.2 and 1.12.2 respectively among many other > > parameters. So far, unfortunately, the matter has not been resolved. > > However, I have noticed that the memory usage on each server > > has built up > > with time after the server being rebooted. I have complained > > about using > > close to 500M even when there were very few calls on the > > server but nobody > > This is a linux question not an asterisk question. I have seen very > recent threads on here which explained what was happening and why, you > should try to review them. > > I think you will find that if you reboot and don't start asterisk, > over-night your memory usage will increase to around 500M. > Most likely, during the night is when your cron scripts run, some of > these traverse the entire filesystem (update/locate) and prompt the OS > to save some of the FS contents in memory (ie cache). > > In any case, it is normal to use almost all of your memory. If you > really have doubts, then look at ps aux which will show you memory usage > per process. > > And, this is unlikely a asterisk problem and more an OS > mis-understanding. > > Regards, > Adam > > -- > Adam Goryachev > Website Managers > Ph: +61 2 9345 4395 [EMAIL PROTECTED] > Fax: +61 2 9345 4396 www.websitemanagers.com.au > > > --__--__-- > > Message: 6 > From: "Asan M." <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Tue, 20 Jan 2004 10:31:25 +0500 > Subject: [Asterisk-Users] Call token is ip$localhost > Reply-To: [EMAIL PROTECTED] > > Hi > In ChangeLog the following is written down: > Asterisk 0.7.1 > -- Fixed timed include context's and GotoIfTime > -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1 > > But all the same where that the bells and vanish as of it gets rid... > == New H.323 Connection created. > -- GK17 is calling host 212.212.212.155 > -- Call token is ip$localhost/6651 > -- Call reference is 6651 > > *CLI> show version > Asterisk CVS-01/19/04-16:49:22 built by..... on a i686 running Linux > > --__--__-- > > Message: 7 > Date: Tue, 20 Jan 2004 00:07:32 -0600 (CST) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP) > Reply-To: [EMAIL PROTECTED] > > Can you clarify this? Does it or doesn't it work? > > bkw > > On Mon, 19 Jan 2004, Asterisk User Group wrote: > > > I had been running an older patched CVS to get VOIP working with NAT and > > everything had been running fine. I just built * on a new box with > > CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. > > Has anything major changed... > > > > [general] > > port = 5060 ; Port to bind to > > bindaddr = 0.0.0.0 ; Address to bind to > > externip = 69.132.68.17 ; Address that we're going to put in SIP > > messages if we're behind a NAT > > localnet = 192.168.1.0 ; Internal NETWORK address > > localmask = 255.255.255.0 ; Internal netmask > > context = default ; Default for incoming calls > > ;srvlookup = yes ; Enable SRV lookups on outbound calls > > ;pedantic = yes ; Enable slow, pedantic checking for > > Pingtel > > ;tos=lowdelay > > ;tos=184 > > ;maxexpirey=3600 ; Max length of incoming registration we > > allow > > ;defaultexpirey=120 ; Default length of incoming/outoing > > registration > > ;notifymimetype=text/plain ; Allow overriding of mime type in > > NOTIFY > > ;videosupport=yes ; Turn on support for SIP video > > disallow=all ; Disallow all codecs > > allow=ulaw ; Allow codecs in order of preference > > allow=ilbc > > > > [1001] > > type=friend > > secret=1001 > > host=dynamic > > username=1001 > > mailbox=1001 > > context=local > > nat=no > > > > [1006] > > type=friend > > secret=oicu812 > > host=dynamic > > username=1006 > > mailbox=1006 > > context=local > > nat=yes > > canreinvite=no > > qualify=500 > > > > Internal SIP users can register it just the outside users. > > > > -gcc > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 8 > Date: Mon, 19 Jan 2004 23:38:25 -0600 > From: Marcin Kuzmicki <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] PLAYBACK multiple files > Reply-To: [EMAIL PROTECTED] > > Cytowanie Charles Hatchette <[EMAIL PROTECTED]>: > > > I'm trying to devise a way to playback more than one file per call when I > > copy my file 'Test.call' into .. var/spool/asterisk/outgoing > > > > Channel: Zap/1/put_your_phone_number_here > > Application: Playback > > Data: demo-thanks + a-second-file + a-third-file > > > > Is there some way to do this? > > Create context in extensions.conf something like > > [myplayback] > exten => s,1,Playback(frist_file) > exten => s,2,Playback(second_file) > ...etc > > and then > use > Context, Extension, and priority to use it > ie. > Channel: Zap/1/put_your_phone_number_here > Context: myplayback > Extension: s > Priority: 1 > > > all above is just a concept not ready copy&paste solution. > > > regards > m. > > > > > --__--__-- > > Message: 9 > Date: Tue, 20 Jan 2004 00:08:26 -0600 (CST) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] user password and call waiting > Reply-To: [EMAIL PROTECTED] > > Use account codes. That works ALOT better. If you require passwords then > look at app_authenticate. > > bkw > > On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote: > > > > > Dear all, > > I have a questions: > > 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those > > phone. I want to be able to log who is using the phones and where to. I'd > > like to use password for each user so that I can keep track who is the > > caller and for how long. > > I read the authenticate application, but I think it is for one user only. > > Forgive my English. > > > > > > Fxo --> phone1 user A use phone1 or phone2 or phone3 after entering > > Fxo --> phone2 password like 1234, so if A want to call from either phones > > Fxo --> phone3 A needs to punch 91234xxxxxxx > > The same with user B, B needs to punch 92345xxxxxx > > And so on. > > But in my logger (either text based or database based), I need to see the > > caller is A and the rest is the same. > > Can I do this with *. What is the effective approach? > > > > 2. I Use digium hardware (FXO and FXS), * v0.5. Can I activate the caller > > waiting feature on the fxs's? > > So if phone 1 is being used, and I called phone 1 from phone 2, phone 1 will > > get call waiting tone, and from phone 2 will hear the connecting tones? > > I put callwaiting=yes in Zapata.conf already. But it didn't work.Any help? > > > > Thanks > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 10 > Date: Tue, 20 Jan 2004 17:20:46 +0100 > From: dkwok <[EMAIL PROTECTED]> > Organization: iware.com.au > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] echo cancellation > Reply-To: [EMAIL PROTECTED] > > This is a cryptographically signed message in MIME format. > > --------------ms020803010300040601040808 > Content-Type: text/plain; charset=us-ascii; format=flowed > Content-Transfer-Encoding: 7bit > > echo cancellation is activated in /etc/asterisk/zapata.conf > > However, how to confirm it? > > Does "zap show channel 1" confirm the existence of echo cancellation? > > -- > David Kwok > > Iaxtel/FWD # 17001813482 > > --------------ms020803010300040601040808 > Content-Type: application/x-pkcs7-signature; name="smime.p7s" > Content-Transfer-Encoding: base64 > Content-Disposition: attachment; filename="smime.p7s" > Content-Description: S/MIME Cryptographic Signature > > MIAGCSqGSIb3DQEHAqCAMIACAQExCzAJBgUrDgMCGgUAMIAGCSqGSIb3DQEHAQAAoIIEvjCC > AlswggHEoAMCAQICARQwDQYJKoZIhvcNAQEEBQAwRDELMAkGA1UEBhMCQVUxDjAMBgNVBAoT > BWl3YXJlMQswCQYDVQQLEwJDQTEYMBYGA1UEAxQPY2FAaXdhcmUuY29tLmF1MB4XDTAzMTEw > MTAwMTIyNFoXDTA0MTAzMTAwMTIyNFowfTELMAkGA1UEBhMCQVUxDDAKBgNVBAgTA05TVzEQ > MA4GA1UEBxMHTlNZRE5FWTEOMAwGA1UEChMFSVdBUkUxGzAZBgNVBAMUEmRrd29rQGl3YXJl > LmNvbS5hdTEhMB8GCSqGSIb3DQEJARYSZGt3b2tAaXdhcmUuY29tLmF1MIGfMA0GCSqGSIb3 > DQEBAQUAA4GNADCBiQKBgQDCcZEUZbESmEA4zQyeVp+t3Q/PU7Mi0tqOnu2BTBWJZ0sv1aRY > bEn1q67fxkQ4Q/x0OWyKv7p7tTZNKF2oSp1TRInCmSleyGOfKm7AR3OSNhYUfGF08vefcl3X > G2Y1nMoXDZUGfas7AbLmKkMgBx0jQ9VKbKzG70ganHgREchvrwIDAQABoyQwIjAgBgNVHREB > Af8EFjAUghJka3dva0Bpd2FyZS5jb20uYXUwDQYJKoZIhvcNAQEEBQADgYEAbGa9xUwYlpja > wMGh/L46YhyolmOqJa4a72sVu1wBVRnVLTXVn7Wc4p7SZaKjTdhOmRS7SmKvm3cPx3u9XKCY > 6nuUrkA9SMAtYuJ9UzE+BMV8/MtC1avEtTZebWjdXy4f8dKc4AVN+WP9YAFGh67a1GmTk6M8 > Ilzm/giua4G18qgwggJbMIIBxKADAgECAgEUMA0GCSqGSIb3DQEBBAUAMEQxCzAJBgNVBAYT > AkFVMQ4wDAYDVQQKEwVpd2FyZTELMAkGA1UECxMCQ0ExGDAWBgNVBAMUD2NhQGl3YXJlLmNv > bS5hdTAeFw0wMzExMDEwMDEyMjRaFw0wNDEwMzEwMDEyMjRaMH0xCzAJBgNVBAYTAkFVMQww > CgYDVQQIEwNOU1cxEDAOBgNVBAcTB05TWURORVkxDjAMBgNVBAoTBUlXQVJFMRswGQYDVQQD > FBJka3dva0Bpd2FyZS5jb20uYXUxITAfBgkqhkiG9w0BCQEWEmRrd29rQGl3YXJlLmNvbS5h > dTCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAwnGRFGWxEphAOM0Mnlafrd0Pz1OzItLa > jp7tgUwViWdLL9WkWGxJ9auu38ZEOEP8dDlsir+6e7U2TShdqEqdU0SJwpkpXshjnypuwEdz > kjYWFHxhdPL3n3Jd1xtmNZzKFw2VBn2rOwGy5ipDIAcdI0PVSmysxu9IGpx4ERHIb68CAwEA > AaMkMCIwIAYDVR0RAQH/BBYwFIISZGt3b2tAaXdhcmUuY29tLmF1MA0GCSqGSIb3DQEBBAUA > A4GBAGxmvcVMGJaY2sDBofy+OmIcqJZjqiWuGu9rFbtcAVUZ1S011Z+1nOKe0mWio03YTpkU > u0pir5t3D8d7vVygmOp7lK5APUjALWLifVMxPgTFfPzLQtWrxLU2Xm1o3V8uH/HSnOAFTflj > /WABRoeu2tRpk5OjPCJc5v4IrmuBtfKoMYICWjCCAlYCAQEwSTBEMQswCQYDVQQGEwJBVTEO > MAwGA1UEChMFaXdhcmUxCzAJBgNVBAsTAkNBMRgwFgYDVQQDFA9jYUBpd2FyZS5jb20uYXUC > ARQwCQYFKw4DAhoFAKCCAWcwGAYJKoZIhvcNAQkDMQsGCSqGSIb3DQEHATAcBgkqhkiG9w0B > CQUxDxcNMDQwMTIwMTYyMDQ2WjAjBgkqhkiG9w0BCQQxFgQUUSYChx4vgx9wJ5RnYFVA795D > K0swUgYJKoZIhvcNAQkPMUUwQzAKBggqhkiG9w0DBzAOBggqhkiG9w0DAgICAIAwDQYIKoZI > hvcNAwICAUAwBwYFKw4DAgcwDQYIKoZIhvcNAwICASgwWAYJKwYBBAGCNxAEMUswSTBEMQsw > CQYDVQQGEwJBVTEOMAwGA1UEChMFaXdhcmUxCzAJBgNVBAsTAkNBMRgwFgYDVQQDFA9jYUBp > d2FyZS5jb20uYXUCARQwWgYLKoZIhvcNAQkQAgsxS6BJMEQxCzAJBgNVBAYTAkFVMQ4wDAYD > VQQKEwVpd2FyZTELMAkGA1UECxMCQ0ExGDAWBgNVBAMUD2NhQGl3YXJlLmNvbS5hdQIBFDAN > BgkqhkiG9w0BAQEFAASBgLpPd814VCfFvCleLVHYKIH6udpQdLBJz7cnCrOa9yhkukFX0nLP > KCLVEgFfsXI67ZagLB1h6F+a++Mirnt5sw5imnC5P19trKFiqtu20SUGx3z9+uPowOd6kAeA > GYyYOqNz6eHCX/F2NywM16cTa5ApRBP6qgjZmdUn8UMCDFHQAAAAAAAA > --------------ms020803010300040601040808-- > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users