Take out the allow=all in your sip.conf and put in allow= for the codec you want to use and disallow=all.
On Wed, 2004-04-07 at 15:18, Roger wrote: > I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone > are registered by the below information > > *CLI> sip show peers > Name/username Host Mask Port Status > 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored > 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored > 2000/2000 192.168.22.198 (D) 255.255.255.255 5060 Unmonitored > > *CLI> sip show users > Username Secret Authen Def.Context A/C > 2002 cisco md5,plaintext demo No > 2001 cisco md5,plaintext demo No > 2000 cisco md5,plaintext demo No > > > All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet. > I've attached my sip.conf and extensions.conf file for review... > > When I start the server and a phone dials another phone I get the below > answer. > > *CLI> -- Executing Dial("SIP/2001-0bb5", "SIP/2002|30|tr") in new stack > -- Called 2002 > -- Got SIP response 488 "Not Acceptable Here" back from 192.168.22.199 > == No one is available to answer at this time > -- Timeout on SIP/2001-0bb5 > > I *believe* the sip response might be from the phone itself - and not a > asterisk misconfig. I'm just wanting a second pair of eyes. > > I put in > > canreinvite=no > > for each phone profile as people have said this is needed for buggy > Cisco phones. > > > ______________________________________________________________________ > ; > ; SIP Configuration for Asterisk > ; > ; Syntax for specifying a SIP device in extensions.conf is > ; SIP/devicename where devicename is defined in a section below. > ; > ; You may also use > ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet > ; (Don't forget to enable DNS SRV records if you want to use this) > ; > ; If you define a SIP proxy as a peer below, you may call > ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] > ; where the proxyhostname is defined in a section below > ; > ; Useful CLI commands to check peers/users: > ; sip show peers Show all SIP peers (including friends) > ; sip show users Show all SIP users (including friends) > ; sip show registry Show status of hosts we register with > ; > ; sip debug Show all SIP messages > ; > > [general] > > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > ;bindaddr = 192.168.22.254 ; 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 > tos = lowdelay ; can be lowdelay, throughput, reliability, mincost > > [2000] > type=friend ; This device takes and makes calls > username=2000 ; Username on device > secret=cisco ; Password for device > ;host=192.168.22.1 ; This host is not on the same IP addr every time > host=dynamic > context=demo ; Inbound calls from this host go here > mailbox=100 ; Activate the message waiting light if this > canreinvite=no > ; voicemailbox has messages in it > > [2001] ; Duplicate of 2000, except with different auth data > type=friend > username=2001 > secret=cisco > host=dynamic > ;host=192.168.22.2 > context=demo > mailbox=101 > canreinvite=no > > [2002] ; Duplicate of 2000, except with different auth data > type=friend > username=2002 > secret=cisco > ;host=192.168.22.3 > host=dynamic > context=demo > mailbox=102 > canreinvite=no > > ______________________________________________________________________ > ; > ; Static extension configuration file, used by > ; the pbx_config module. This is where you configure all your > ; inbound and outbound calls in Asterisk. > ; > [incoming] > exten => s,1,Echo ;for testing the connection > ;exten => s,1,Playback,demo-thanks ;for playing a file > ; > ; The "General" category is for certain variables. > ; > [general] > ; > ; If static is set to no, or omitted, then the pbx_config will rewrite > ; this file when extensions are modified. Remember that all comments > ; made in the file will be lost when that happens. > ; > ; XXX Not yet implemented XXX > ; > static=yes > ; > ; if static=yes and writeprotect=no, you can save dialplan by > ; CLI command 'save dialplan' too > ; > writeprotect=no > > ; You can include other config files, use the #include command (without the ';') > ; Note that this is different from the "include" command that includes contexts > within > ; other contexts. The #include command works in all asterisk configuration files. > ;#include "filename.conf" > > ; The "Globals" category contains global variables that can be referenced > ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable > ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid > ; > [globals] > CONSOLE=Console/dsp ; Console interface for demo > ;CONSOLE=Zap/1 > ;CONSOLE=Phone/phone0 > IAXINFO=guest ; IAXtel username/password > ;IAXINFO=myuser:mypass > TRUNK=Zap/g2 ; Trunk interface > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > ;TRUNK=IAX2/user:[EMAIL PROTECTED] > > ; > ; Any category other than "General" and "Globals" represent > ; extension contexts, which are collections of extensions. > ; > ; Extension names may be numbers, letters, or combinations > ; thereof. If an extension name is prefixed by a '_' > ; character, it is interpreted as a pattern rather than a > ; literal. In patterns, some characters have special meanings: > ; > ; X - any digit from 0-9 > ; Z - any digit from 1-9 > ; N - any digit from 2-9 > ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) > ; . - wildcard, matches anything remaining (e.g. _9011. matches > ; anything starting with 9011 excluding 9011 itself) > ; > ; For example the extension _NXXXXXX would match normal 7 digit dialings, > ; while _1NXXNXXXXXX would represent an area code plus phone number > ; preceeded by a one. > ; > ; Contexts contain several lines, one for each step of each > ; extension, which can take one of two forms as listed below, > ; with the first form being preferred. One may include another > ; context in the current one as well, optionally with a > ; date and time. Included contexts are included in the order > ; they are listed. > ; > ;[context] > ;exten => someexten,priority,application(arg1,arg2,...) > ;exten => someexten,priority,application,arg1|arg2... > ; > ; Timing list for includes is > ; > ; <time range>|<days of week>|<days of month>|<months> > ; > ;include => daytime|9:00-17:00|mon-fri|*|* > ; > ; ignorepat can be used to instruct drivers to not cancel dialtone upon > ; receipt of a particular pattern. The most commonly used example is > ; of course '9' like this: > ; > ;ignorepat => 9 > ; > ; so that dialtone remains even after dialing a 9. > ; > > ; > ; Here are the entries you need to participate in the IAXTEL > ; call routing system. Most IAXTEL numbers begin with 1-700, but > ; there are exceptions. For more information, and to sign > ; up, please go to www.gnophone.com or www.iaxtel.com > ; > [iaxtel700] > exten => _91700NXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) > > [iaxprovider] > ;switch => IAX2/user:[EMAIL PROTECTED]/mycontext > > [trunkint] > ; > ; International long distance through trunk > ; > exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _9011.,2,Congestion > > [trunkld] > ; > ; Long distance context accessed through trunk > ; > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91NXXNXXXXXX,2,Congestion > > [trunklocal] > ; > ; Local seven-digit dialing accessed through trunk interface > ; > exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _9NXXXXXX,2,Congestion > > [trunktollfree] > ; > ; Long distance context accessed through trunk interface > ; > exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91800NXXXXXX,2,Congestion > exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91888NXXXXXX,2,Congestion > exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91877NXXXXXX,2,Congestion > exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91866NXXXXXX,2,Congestion > > [international] > ; > ; Master context for international long distance > ; > ignorepat => 9 > include => longdistance > include => trunkint > > [longdistance] > ; > ; Master context for long distance > ; > ignorepat => 9 > include => local > include => trunkld > > [local] > ; > ; Master context for local, toll-free, and iaxtel calls only > ; > ignorepat => 9 > include => default > include => parkedcalls > include => trunklocal > include => iaxtel700 > include => trunktollfree > include => iaxprovider > ; > ; You can use an alternative switch type as well, to resolve > ; extensions that are not known here, for example with remote > ; IAX switching you transparently get access to the remote > ; Asterisk PBX > ; > ; switch => IAX2/user:[EMAIL PROTECTED]/local > > [macro-stdexten]; > ; > ; Standard extension macro: > ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well > ; ${ARG2} - Device(s) to ring > ; > exten => s,1,Dial(${ARG2},20) ; Ring the interface, > 20 seconds maximum > exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send > to voicemail w/ unavail announce > exten => s,3,Goto(default,s,1) ; If they > press #, return to start > exten => s,102,Voicemail(b${ARG1}) ; If busy, send to > voicemail w/ busy announce > exten => s,103,Goto(default,s,1) ; If they press #, > return to start > > > [demo] > ; > ; We start with what to do when a call first comes in. > ; > exten => s,1,Wait,1 ; Wait a second, just for fun > exten => s,2,Answer ; Answer the line > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds > exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message > exten => s,6,BackGround(demo-instruct) ; Play some instructions > > exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. > exten => 2,2,Goto(s,6) > > exten => 3,1,SetLanguage(fr) ; Set language to french > exten => 3,2,Goto(s,5) ; Start with the congratulations > > exten => 1000,1,Goto(default,s,1) > ; > ; We also create an example user, 1234, who is on the console and has > ; voicemail, etc. > ; > exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." > ; (but skip if channel is not up) > exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) > > exten => 1235,1,Voicemail(u1234) ; Right to voicemail > > exten => 1236,1,Dial(Console/dsp) ; Ring forever > exten => 1236,2,Voicemail(u1234) ; Unless busy > > exten => 2000,1,Dial(SIP/2000,30,tr) > > exten => 2001,1,Dial(SIP/2001,30,tr) > > exten => 2002,1,Dial(SIP/2002,30,tr) > > ; > ; # for when they're done with the demo > ; > exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" > exten => #,2,Hangup ; Hang them up. > > ; > ; A timeout and "invalid extension rule" > ; > exten => t,1,Goto(#,1) ; If they take too long, give up > exten => i,1,Playback(invalid) ; "That's not valid, try again" > > ; > ; Create an extension, 500, for dialing the > ; Asterisk demo. > ; > exten => 500,1,Playback(demo-abouttotry); Let them know what's going on > exten => 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the Asterisk > demo > exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site > exten => 500,4,Goto(s,6) ; Return to the start over message. > > ; > ; Create an extension, 600, for evaulating echo latency. > ; > exten => 600,1,Playback(demo-echotest) ; Let them know what's going on > exten => 600,2,Echo ; Do the echo test > exten => 600,3,Playback(demo-echodone) ; Let them know it's over > exten => 600,4,Goto(s,6) ; Start over > > ; > ; Give voicemail at extension 8500 > ; > exten => 8500,1,VoicemailMain > exten => 8500,2,Goto(s,6) > ; > ; Here's what a phone entry would look like (IXJ for example) > ; > ;exten => 1265,1,Dial(Phone/phone0,15) > ;exten => 1265,2,Goto(s,5) > > ;[mainmenu] > ; > ; Example "main menu" context with submenu > ; > ;exten => s,1,Answer > ;exten => s,2,Background(thanks) ; "Thanks for calling press 1 for > sales, 2 for support, ..." > ;exten => 1,1,Goto(submenu,s,1) > ;exten => 2,1,Hangup > ;include => default > ; > ;[submenu] > ;exten => s,1,Ringing ; Make them comfortable with 2 > seconds of ringback > ;exten => s,2,Wait,2 > ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales department. > Press 1 for steve, 2 for..." > ;exten => 1,1,Goto(default,steve,1) > ;exten => 2,1,Goto(default,mark,2) > > [default] > ; > ; By default we include the demo. In a production system, you > ; probably don't want to have the demo there. > ; > include => demo > > > ; Real extensions would go here. Generally you want real extensions to be 4 or 5 > ; digits long (although there is no such requirement) and start with a single > ; digit that is fairly large (like 6 or 7) so that you have plenty of room to > ; overlap extensions and menu options without conflict. You can alias them with > ; names, too and use global variables > > > ;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is > something like Zap/2 > ;exten => mark,1,Goto(6275|1) ; alias mark > to 6275 > ;exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil > ;exten => wil,1,Goto(6236|1) > ; > ; Some other handy things are an extension for checking voicemail via > ; voicemailmain > ; > ;exten => 8500,1,VoicemailMain > ;exten => 8500,2,Hangup > ; > ; Or a conference room (you'll need to edit meetme.conf to enable this room) > ; > ;exten => 8600,1,Meetme,1234 > ; > ; Or playing an announcement to the called party, as soon it answers > ; > ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg)) > ; > ; For more information on applications, just type "show applications" at your > ; friendly Asterisk CLI prompt. > ; > ; 'show application <command>' will show details of how you > ; use that particular application in this file, the dial plan. > ; -- 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." _______________________________________________ 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