Jain, Sonal
2004-Apr-08 12:42 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3373 - 14 msgs
Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, April 08, 2004 3:30 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3373 - 14 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com 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 asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: can't hear vm audio (Mark Phillips) 2. Re: Zapata required? (Mark Phillips) 3. Re: TigerJet ISDN card (Michael Welter) 4. Using Skinny with a 7905G phone (Chris Barnett) 5. Re: Asterisk & 3com nbx 100 support (Eric Wieling) 6. Re: Siemens EWSD 13 (CW_ASN) 7. Auto Attendant?? (James Moran) 8. Re: TigerJet ISDN card (Michael Welter) 9. FreeBSD port of asterisk (David W. Chapman Jr.) 10. TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support (Kyle Hagan) 11. Re: RE: RxFax/spandsp: not disconnecting (Derek) 12. Local Calling Area database? (Scott Laird) 13. Re: ADPCM 4-bit, 6 kHz (Steven Critchfield) 14. RE: H.323 Seg faulting (Derek Samford) --__--__-- Message: 1 Date: Thu, 8 Apr 2004 13:03:59 -0400 (EDT) Subject: Re: [Asterisk-Users] can't hear vm audio From: "Mark Phillips" <kc2eni@nyc-ares.org> To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com OK, this is too damn freaky! Now it work but I didn't do anything. Actually, that's not quite true. I found that if I do modprobe zaptel then modprobe wcfxo then ztcfg the vm audio works. If I do service zaptel start it doesn't. What gives?> So I've been fighting to get the X100P working. A battle which I've kinda > won but not without a cost. > > Before I won the Zaptel battle I was able to hear all of the messages that > asterisk plays. For example, when I'm accessing VoiceMail I would have > been requested to input my password. This did work but now it doesn't. > Asterisk does show that it is playing the file but no audio is heard. > > I have audio on regular SIP based calls as well as IAX based ones. I'me > not getting and audio when I make a ZAP call. > > Ideas? > > Mark > > > > G7LTT/KC2ENI > Mark Phillips > > > > G7LTT/KC2ENI > Mark Phillips > _______________________________________________ > 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 >G7LTT/KC2ENI Mark Phillips --__--__-- Message: 2 Date: Thu, 8 Apr 2004 13:05:06 -0400 (EDT) Subject: Re: [Asterisk-Users] Zapata required? From: "Mark Phillips" <kc2eni@nyc-ares.org> To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com You did unhash the ztdummy in the Makefile before compiling it right?> Steven Kokinos wrote: > >>Ho do I go about loading the ztdummy driver after unloading zap? >> >> >> > $ su - > # modprobe ztdummy > > >>Thanks, >> >>-Steve >> >> >> > > _______________________________________________ > 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 >G7LTT/KC2ENI Mark Phillips --__--__-- Message: 3 Date: Thu, 08 Apr 2004 11:23:31 -0600 From: Michael Welter <mike@introspect.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TigerJet ISDN card Reply-To: asterisk-users@lists.digium.com In Linux config, CAPI2.0 support is under the "Active ISDN Cards" category. Does this mean CAPI will not work with static cards? Thanks, Mark Phillips wrote:> Is it CAPI compliant? if so yes > > > >>Is there any Linux/* support for the TigerJet ISDN card? >> >>-brian >> > > > > G7LTT/KC2ENI > Mark Phillips > _______________________________________________ > 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 > >-- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com --__--__-- Message: 4 Date: Thu, 8 Apr 2004 18:31:07 +0100 From: "Chris Barnett" <chris.barnett@insphire.com> To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] Using Skinny with a 7905G phone Reply-To: asterisk-users@lists.digium.com Hi All, I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk server but I'm having problems getting the phone to answer a call or make a call. I'm using the stable branch of the asterisk CVS on a RH9 box. I have got the phone to register with * and it retrieves it's extension number, date & time etc., but when I pick the handset up it's just dead silence (but I get tones when pressing the numbers on the keypad). There are no debug messages or otherwise shown on the * console (even with very verbose turned on and skinny debugging turned on). If I ring the phone from a SIP PC softphone (X-Ten lite) the phone rings but continues to ring even when the handset is picked up - until the 20 seconds timeout occurs and the call is transferred to the extensions voicemail box. When the call is made I do see debug messages telling me the call is being made and sent to the skinny extension. I have experimented with allowing and disallowing various formats to no avail. I'm sure I've missed something out but after many days and hours searching every resource I can find I haven't been able to progress any further - any ideas what I've done wrong or missed out - or is what I am trying to achieve simply not possible at the moment. Below is my skinny.conf; [general] port =3D 2000 Bindaddr =3D 0.0.0.0 dateFormat =3D D-M-Y keepAlive =3D 120 ; allow =3D all ; disallow =3D [Ext104] Device=3DSEP000E386DCF06 Version=3DP021114C ; version number from the phone Nat=3D0 Callerid=3D"Cisco IP Phone" <104> Mailbox=3D1004 Callwaiting=3D1 Transfer=3D1 Threewaycalling=3D1 Context=3Ddefault Line =3D> 104 Host=3D192.168.1.55 And this is the section from extensions.conf; <snip> Exten =3D> 104,1,Dial(Skinny/104@Ext104,20,tr) Exten =3D> 104,2,VoiceMail,u104 Exten =3D> 104,102,VoiceMail,b104 <snip> Many thanks, Chris Barnett --__--__-- Message: 5 Subject: Re: [Asterisk-Users] Asterisk & 3com nbx 100 support From: Eric Wieling <eric@fnords.org> To: asterisk-users@lists.digium.com Organization: BTEL Consulting Date: Thu, 08 Apr 2004 12:33:38 -0500 Reply-To: asterisk-users@lists.digium.com Try www.google.com "site:lists.digium.com NBX" without the quotes, of course. There's also a google search box on the same page you use to sign up for the mailing lists, but it never worked right for me. On Thu, 2004-04-08 at 11:59, Jeremy Koski wrote:> No. The only way I found to search the archives was month by month. > > There's an option to download all of the mailing list archives, but its > 129MB... > > On Thu, 8 Apr 2004, Eric Wieling wrote: > > > On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote: > > > Does anybody know if the 3com NBX 100 phones will work with > > > Asterisk? The phones require a boot image to be sent either through > > > layer2 or layer3 before they will function properly after being powered > > > on each time. > > > > You didn't see the information when you searched the mailing list > > archives?-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." --__--__-- Message: 6 From: "CW_ASN" <cw_asn@fibertel.com.ar> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Siemens EWSD 13 Date: Thu, 8 Apr 2004 14:40:26 -0300 Reply-To: asterisk-users@lists.digium.com In fact, with EWSD V13 you can't remove CRC4 in PRI mode. Regards, Gus ----- Original Message ----- From: "Storer, Darren" <starusers@comgate.tv> To: <asterisk-users@lists.digium.com> Sent: Wednesday, April 07, 2004 8:32 PM Subject: RE: [Asterisk-Users] Siemens EWSD 13> Hi, > > I had exactly the same symptoms today with a co-located * connected to a > Public Switch here in the UK. The problem was solved by insisting that the > Telco turned on CRC4 at their end and then, after an 'init 6', layer two > settled down on both systems. > > I was taught that if you are connecting to a full specification Q.931 > circuit, CRC4 should be enabled by default; in the event that one end does > not support CRC4 the other end should auto-negotiate back and the circuit > should still align without problems. Having said all of this I have yet to > see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and > suspect that I was not told the truth in the first place... > > Selection of CRC4 seems to be random from Telco to Telco even on aninstall> by install basis within the same Carrier. It's the first thing to checkwhen> new kit appears to be unstable.. > > HTH > > Darren > -- > Comgate > Telco>Internet<Broadcast > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of > asterisk@geek.be > Sent: 07 April 2004 14:59 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Siemens EWSD 13 > > > Hi all, > > Has anyone got any experience with hooking Asterisk up with a > Siemens EWSD 13 switch over a E1/PRI ? > We're located in Belgium (Europe) and one of our telecom partners > uses this switch. > > We connected one of our TE410P ports with their switch, but the status > light on the TE410P card keeps blinking red. > On their side they are getting a DSA (distance service alarm) error, so > this normally means the devices 'see' eachother.. but there are still > problems with the signalling. > > Our config below is the same as we are using for MCI, one of our other > telecom partners. > > We tried changing the LBO and timing, but no luck. > As you see the signalling is carried over channel 16 (default). > > TX and RX have also been regularly switched, so no luck.. > > Their switch is providing the timing. > > The telecom operator has double checked the asterisk config several > times, and it's conform to their setup. > > The only thing they couldn't find in the Asterisk config is a > 'multiframing' option. But I presume this is automatically detected or > set by default ? > They also tried normal/single(?) framing, but no difference. > > The card has also been tested with our MCI E1, and works flawlessly, so > no hardware issue. > > Anyone got any further ideas ? > > Any info or help greatly appreciated! > > Our config, > > *** zaptel.conf *** > span=1,1,6,ccs,hdb3,crc4,yellow > bchan=1-15 > bchan=17-31 > dchan=16 > > *** zapata.conf *** > [channels] > switchtype=euroisdn > signalling=pri_cpe > pridialplan=unknown > > group=1 > channel => 1-15,17-31 > > <other zapata standard config> > > > > _______________________________________________ > 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 > > _______________________________________________ > 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--__--__-- Message: 7 From: James Moran <jmoran@potentialtech.com> To: Asterisk <asterisk-users@lists.digium.com> Organization: Potential Technologies Date: Thu, 08 Apr 2004 13:48:24 -0400 Subject: [Asterisk-Users] Auto Attendant?? Reply-To: asterisk-users@lists.digium.com I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some??? --__--__-- Message: 8 Date: Thu, 08 Apr 2004 11:51:56 -0600 From: Michael Welter <mike@introspect.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TigerJet ISDN card Reply-To: asterisk-users@lists.digium.com In Linux config, CAPI2.0 support is under the "Active ISDN Cards" category. Does this mean CAPI will not work with _passive_ cards? Michael Welter wrote:> In Linux config, CAPI2.0 support is under the "Active ISDN Cards" > category. Does this mean CAPI will not work with static cards? > > Thanks, > > Mark Phillips wrote: > >> Is it CAPI compliant? if so yes >> >> >> >>> Is there any Linux/* support for the TigerJet ISDN card? >>> >>> -brian >>> >> >> >> >> G7LTT/KC2ENI >> Mark Phillips >> _______________________________________________ >> 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 >> >> >-- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com --__--__-- Message: 9 Date: Thu, 8 Apr 2004 12:55:11 -0500 From: "David W. Chapman Jr." <dwcjr@inethouston.net> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FreeBSD port of asterisk Reply-To: asterisk-users@lists.digium.com In our FreeBSD port of Asterisk, we have a lot of local patches and I was wondering if it were possible to get some of them merged into the Asterisk source base. Thanks -- David W. Chapman Jr. dwcjr@inethouston.net Raintree Network Services, Inc. <www.inethouston.net> --__--__-- Message: 10 From: "Kyle Hagan" <khagan@nuvoinc.com> To: <asterisk-users@lists.digium.com> Date: Thu, 8 Apr 2004 10:54:22 -0700 Subject: [Asterisk-Users] TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support Reply-To: asterisk-users@lists.digium.com Finally got my Asterisk (Test) system up and working with a TDM Starter kit. WOOHOO!!! Goona be buying the T1 cards soon to fully implement the system and FINALLY get rid of our old Fujitsu Starlog POS. Was just wondering about the Fax Support. Features say its not complete. Any idea when this could be available? Kyle --__--__-- Message: 11 From: Derek <derek@incrediblefresh.com> Subject: Re: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting Date: Thu, 8 Apr 2004 14:02:27 -0400 To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com I installed rxfax and spandsp. Everything looks good except when receiving a fax, only the first page is captured, even though the sending machine seems to be transmitting all pages. Am I doing something wrong? -D Derek Irwin Information Technology IncredibleFresh Inc. Naples, FL "I've been booting my Windows box daily for the last year. The computer's OK, but my shoe's starting to wear out." On Mar 31, 2004, at 11:01 AM, Steve Underwood wrote:> Reynaldo Simbulan wrote: > >> Hi Steve, >> >> I am having this problem in which RxFax is still holding the file >> after >> receiving a complete fax. Somehow the zap channel is still active but >> on the >> fax client it was sent successfully. >> If you call the line it is still busy. >> >> > spandsp-0.0.1k.tar.gz and updated app_rxfax.c and app_txfax.c files > are available for download. They address this disconnect issue, and > have a few other minor tweaks. > > Regards, > Steve > > _______________________________________________ > 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 >--__--__-- Message: 12 To: asterisk-users@lists.digium.com From: Scott Laird <scott@sigkill.org> Date: Thu, 8 Apr 2004 11:04:48 -0700 Subject: [Asterisk-Users] Local Calling Area database? Reply-To: asterisk-users@lists.digium.com Is there an easy way to get information about local calling areas out of telcos? I'm trying to get a list of area codes and prefixes in my local calling area out of Verizon, and it looks like they've stopped providing the information online. Is there an easy source that I'm missing, or do I need to call them and have them mail me a copy every few months? Scott --__--__-- Message: 13 Subject: Re: [Asterisk-Users] ADPCM 4-bit, 6 kHz From: Steven Critchfield <critch@basesys.com> To: asterisk-users@lists.digium.com Date: Thu, 08 Apr 2004 13:20:15 -0500 Reply-To: asterisk-users@lists.digium.com On Tue, 2004-04-06 at 10:49, Steve Underwood wrote:> Why do people get this uncontrollable urge to post, when the don't know > the correct answer? :-)Having the absolute correct answer isn't always important if it steers the requester in the right direction of self enlightenment. Don't discourage new users too much from answering questions as we need new blood to carry on the fight of educating the even newer users. I know I for one have been pretty burnt out on it and skip all but a small handful of questions a week. -- Steven Critchfield <critch@basesys.com> --__--__-- Message: 14 Subject: RE: [Asterisk-Users] H.323 Seg faulting Date: Thu, 8 Apr 2004 14:25:49 -0400 From: "Derek Samford" <dsamford@netphoneblue.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com This is a multi-part message in MIME format. ------_=_NextPart_001_01C41D96.EAEAE4F8 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable I've placed a bounty on my bug. See http://bugs.digium.com/bug_view_page.php?bug_id=3D0001334 =20 =20 =20 _____ =20 From: Derek Samford=20 Sent: Wednesday, April 07, 2004 4:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] H.323 Seg faulting =20 Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? =20 http://bugs.digium.com/bug_view_page.php?bug_id=3D0001381 =20 =20 Thanks, Derek ------_=_NextPart_001_01C41D96.EAEAE4F8 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-microsoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; charset=3Dus-ascii"> <meta name=3DGenerator content=3D"Microsoft Word 11 (filtered medium)"> <!--[if !mso]> <style> v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} </style> <![endif]--> <style> <!-- /* Font Definitions */ @font-face {font-family:Tahoma; panose-1:2 11 6 4 3 5 4 4 2 4;} /* Style Definitions */ p.MsoNormal, li.MsoNormal, div.MsoNormal {margin:0in; margin-bottom:.0001pt; font-size:12.0pt; font-family:"Times New Roman";} a:link, span.MsoHyperlink {color:blue; text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed {color:purple; text-decoration:underline;} span.EmailStyle17 {mso-style-type:personal; font-family:Arial; color:windowtext;} span.EmailStyle18 {mso-style-type:personal-reply; font-family:Arial; color:navy;} @page Section1 {size:8.5in 11.0in; margin:1.0in 1.25in 1.0in 1.25in;} div.Section1 {page:Section1;} --> </style> </head> <body lang=3DEN-US link=3Dblue vlink=3Dpurple> <div class=3DSection1> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>I’ve placed a bounty on my bug. See </span></font><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'><a href=3D"http://bugs.digium.com/bug_view_page.php?bug_id=3D0001334">http://bugs.digium.com/bug_view_page.php?bug_id=3D0001334</a><o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p> <div> <div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'> <hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1> </span></font></div> <p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt; font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Derek Samford <br> <b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, April 07, 2004 4:26 PM<br> <b><span style=3D'font-weight:bold'>To:</span></b> asterisk-users@lists.digium.com<br> <b><span style=3D'font-weight:bold'>Subject:</span></b> [Asterisk-Users] H.323 Seg faulting</span></font><o:p></o:p></p> </div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'>Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault?<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'><a href=3D"http://bugs.digium.com/bug_view_page.php?bug_id=3D0001381">http://bugs.digium.com/bug_view_page.php?bug_id=3D0001381</a><o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'>Thanks,<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10.0pt; font-family:Arial'>Derek<o:p></o:p></span></font></p> </div> </body> </html> ------_=_NextPart_001_01C41D96.EAEAE4F8-- --__--__-- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest
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