Rich Adamson
2003-Sep-07 10:04 UTC
[Asterisk-Users] New cvs compile; basic operational question, please.
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
Rich Adamson
2003-Sep-07 11:16 UTC
[Asterisk-Users] New cvs compile; basic operational question, please.
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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users---------------End of Original Message-----------------
Paul Cheng
2003-Sep-07 11:57 UTC
[Asterisk-Users] New cvs compile; basic operational question, please.
Make sure that both phones are set to accept the same codecs. The Not Acceptable Here is usually when the SIP negotiation fails for a common codec. Use SIP DEBUG at the CLI to see the transcripts for details. You might want to use in sip.conf allow and disallow statements for codecs as well. On Sunday, September 7, 2003, at 07:04 PM, Rich Adamson wrote:> 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >