I downloaded it yesterday,first stuck by the login name and password in its admin page,there is no place in the document mentioned the password.It took me half hour to google,finally found the login name is maint,felt quite frustrated,why they lost such important info.? then I added some accounts according to the instruction ,seemed quite easy.again ,encountered several problems unsolved after 2 hours spent.I have 3 sip UAs,one ht100,2 xten in 2 pcs.Only ht100 can make outgoing calls,2 softphones showed "REGISTERED" and can receive incoming call,but failed to make outgoing call.always showing "call failed 407,authentication reqired".In the caes ht100 made outgoing call to one softphone,ht100 received cristal clear sound,but softphone received sound bearly hearable. my pc which runing ASTERISK is athlon 1G 256g ram I am wondering if this thing really work ,anyone can give suggestion? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
Kerry Garrison
2005-Mar-25 18:25 UTC
[Asterisk-Users] Does asterisk@home 0.6 really work???
Yes it really works, I have some how-to tips at http://www.geekgazette.com. -Kerry -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of sf sf Sent: Friday, March 25, 2005 5:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Does asterisk@home 0.6 really work??? I downloaded it yesterday,first stuck by the login name and password in its admin page,there is no place in the document mentioned the password.It took me half hour to google,finally found the login name is maint,felt quite frustrated,why they lost such important info.? then I added some accounts according to the instruction ,seemed quite easy.again ,encountered several problems unsolved after 2 hours spent.I have 3 sip UAs,one ht100,2 xten in 2 pcs.Only ht100 can make outgoing calls,2 softphones showed "REGISTERED" and can receive incoming call,but failed to make outgoing call.always showing "call failed 407,authentication reqired".In the caes ht100 made outgoing call to one softphone,ht100 received cristal clear sound,but softphone received sound bearly hearable. my pc which runing ASTERISK is athlon 1G 256g ram I am wondering if this thing really work ,anyone can give suggestion? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ _______________________________________________ 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
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of sf sf > Sent: Friday, March 25, 2005 8:14 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Does asterisk@home 0.6 really work??? > > I downloaded it yesterday,first stuck by the login name and passwordin> its > admin page,there is no place in the document mentioned the password.It > took > me half hour to google,finally found the login name is maint,feltquite> frustrated,why they lost such important info.? > then I added some accounts according to the instruction ,seemed quite > easy.again ,encountered several problems unsolved after 2 hoursspent.I> have > 3 sip UAs,one ht100,2 xten in 2 pcs.Only ht100 can make outgoingcalls,2> softphones showed "REGISTERED" and can receive incoming call,butfailed to> make outgoing call.always showing "call failed 407,authentication > reqired".In the caes ht100 made outgoing call to one softphone,ht100 > received cristal clear sound,but softphone received sound bearlyhearable.> > my pc which runing ASTERISK is athlon 1G 256g ram > > I am wondering if this thing really work ,anyone can give suggestion? >Works great for me. Running several VoIP accounts, SIP phone, house phones and much more. Maybe you should start investing a little more time in honing your search skills. Five seconds on the Wiki I found the following: http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home Tells the default root password and the default maint password for the AMP login. Also on the same page is a link to the Asterisk@Home main page which has a handbook and a download link which take you to Source Forge where there is a forum you can ask questions about Asterisk@Home. Robert
Hello,
I am trying to setup spandsp for the first time. I have spandsp
0.0.2pre18 and * v1.0.8 patched. For testing I have this extension:
exten => 1234,1,rxfax(/tmp/testfax.tif)
* answers the call, but makes no fax tones. I get this logged:
-- Executing RxFAX("SIP/x.x.x.x-00668660",
"/tmp/testfax.tif") in
new stack
then nothing, silence, no communication.
Any help is appreciated.
Ted
I got this working on Fedora Core 3 i386, but it does not work on CentOS x86_64. Anything I should know about x86_64? Thanks. Ted Theodore Cekan wrote:> Hello, > > I am trying to setup spandsp for the first time. I have spandsp > 0.0.2pre18 and * v1.0.8 patched. For testing I have this extension: > > exten => 1234,1,rxfax(/tmp/testfax.tif) > > * answers the call, but makes no fax tones. I get this logged: > > -- Executing RxFAX("SIP/x.x.x.x-00668660", "/tmp/testfax.tif") in > new stack > > then nothing, silence, no communication. > > Any help is appreciated. > > Ted > _______________________________________________ > 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
Theodore Cekan wrote:> I got this working on Fedora Core 3 i386, but it does not work on > CentOS x86_64. Anything I should know about x86_64?I have reports from people who say spandsp works OK on x86_64. I had to do a few fixed for them to make it build, which are in any recent copy of spandsp. I just acquired a new X2 machine, so I will be doing more extensive testing of my software built for x86_64 over the next few weeks. Regards, Steve> Thanks. > > Ted > > > Theodore Cekan wrote: > >> Hello, >> >> I am trying to setup spandsp for the first time. I have spandsp >> 0.0.2pre18 and * v1.0.8 patched. For testing I have this extension: >> >> exten => 1234,1,rxfax(/tmp/testfax.tif) >> >> * answers the call, but makes no fax tones. I get this logged: >> >> -- Executing RxFAX("SIP/x.x.x.x-00668660", "/tmp/testfax.tif") in >> new stack >> >> then nothing, silence, no communication. >> >> Any help is appreciated. >> >> Ted >> _______________________________________________ >> 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 >