John, Excellent, I removed them and works fine. I started playing with MOH, but haven't seen an example of how to specify this on a per system or per extension (but I haven't googled for it yet either). I do have one line uncommented in the moh config file, but I assume I need to do something in sip.conf or extensions.conf to make it work.
Suggestions? Rich ------------------------ > Rich - > Leave out the "allow" lines entirely, including the "allow=all" - > this was a problem I discovered post-publishing (that I thought I > corrected in the notes, but I see that it's not there.) The "allow=" > lines need a "disallow=" line to balance them. If you leave both > out, the system will choose the "right" codec, but if you only put > one in, things get twisted up a bit. I've updated the article > (again?) > > JT > > > > >Just stumbled across the problem noted in my original post below. I added: > >allow=ulaw > >allow=ilbc > > > >to sip.conf instead of the recommended 'allow=all' and now all phones work. > > > >Can someone help me understand this? (It would appear, based on my very much > >lack of experience, that * was attempting to set up the conversation > >using g723, > >when all of the phones have 'default=ulaw' definitions. Should I leave the > >ulaw definition for future production use, or is this really something that > >I did to read/learn more about for a very small office use?) > > > >Rich > > > >------------------------ > >> Can someone offer a hint on what I'm doing wrong with the basic * config? > >> > >> Just implemented * for the first time using yesterday's cvs. The initial > > > configs are based on John Todd's article at > >http://www.onlamp.com/lpt/a/3956, > >> and using two 7960's for initial testing. When one 7960 calls the other, I > >> get the following and the call is dropped: > >> > >> -- Executing Dial("SIP/3001-ec1c", "SIP/3000|20") in new stack > >> -- Called 3000 > >> -- Got SIP response 488 "Not Acceptable Here" back from 206.222.193.92 > >> == No one is available to answer at this time > >> -- Executing VoiceMail("SIP/3001-ec1c", "u3000") in new stack > >> == Parsing '/etc/asterisk/voicemail.conf': Found > >> -- Playing 'vm/3000/unavail' > >> > >> My sip.conf looks like: > >> [general] > >> > >> port = 5060 ; Port to bind to (SIP is 5060) > >> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > >> allow=all ; Allow all codecs > >> context = bogon-calls ; Send SIP callers that we don't know about here > >> > >> [3000] > >> type=friend ; This device takes and makes calls > >> username=3000 ; Username on device > >> secret=npi2003 ; Password for device > >> host=dynamic ; This host is not on the same IP addr every time > >> context=from-sip ; Inbound calls from this host go here > >> mailbox=100 ; Activate the message waiting light if this > >> ; voicemailbox has messages in it > >> > >> [3001] > >> type=friend ; This device takes and makes calls > >> username=3001 ; Username on device > >> secret=npi2003 ; Password for device > > > host=dynamic ; This host is not on the same IP addr every time > > > context=from-sip ; Inbound calls from this host go here > > > mailbox=100 ; Activate the message waiting light if this > >> --------------------------------------- > >> > >> and my extensions.conf looks like: > >> > >> [general] > >> static=yes ; These two lines prevent the command-line interface > >> writeprotect=yes ; from overwriting the config file. Leave them here. > >> > >> [bogon-calls] > >> exten => _.,1,Congestion > >> > >> [from-sip] > >> exten => 3000,1,Dial(SIP/3000,20) > >> exten => 3000,2,Voicemail(u3000) > >> exten => 3000,102,Voicemail(b3000) > >> exten => 3000,103,Hangup > >> > >> exten => 3001,1,Dial(SIP/3001,20) > >> exten => 3001,2,Voicemail(u3001) > > > exten => 3001,102,Voicemail(b3001) > >> exten => 3001,103,Hangup > >> > >> exten => 3999,1,VoicemailMain(${CALLERIDNUM}) > >> > >> Apparently I'm doing something wrong, but since this is my first attempt > >> at making * work, I don't actually have a clue what I'm doing (yet). > >> > >> Asterisk did complile and install clean the first time (on new RH9 system), > >> and both 7960's are registered. In some attempts to dial, I do receive > >> vmail announcements, etc, so whatever I've done wrong I'm guessing it must > >> be in the above config statements. > >> > >> If someone would kindly point out my error (and maybe a constructive comment > >> about the error so I can learn), if would be greatly appreciated. > >> > >> TIA, > >> Rich > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > >---------------End of Original Message----------------- > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
