hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ -------------- next part -------------- ; ; Static extension configuration file, used by ; the pbx_config module. This is where you configure all your ; inbound and outbound calls in Asterisk. ; ; ; 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] [sip] exten => 100,1,Dial(SIP/100) exten => 150,1,Dial(SIP/150) exten => 200,1,Dial(SIP/200) exten => 200,1,Dial(SIP/250) -------------- next part -------------- ; ; 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/username@domain 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/user@proxyhostname ; 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] context=sip ; Default context for incoming calls ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) realm=home.net ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=192.168.0.2 ; IP address to bind to (0.0.0.0 binds to all) [100] type=friend username=100 secret=100 fromuser=100 host=dynamic context=sip dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info progressinband=no ; Polycom phones don't work properly with "never" [150] type=friend username=150 secret=150 fromuser=150 host=dynamic context=sip dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info [200] type=friend username=200 fromuser=200 secret=200 host=dynamic context=sip dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info progressinband=no ; Polycom phones don't work properly with "never" [250] type=friend username=250 fromuser=250 secret=250 host=dynamic context=sip dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info progressinband=no ; Polycom phones don't work properly with "never"
Hi Harry -> I try to set up two lines per ip 300 phone, > registration is ok but i get Failure to authenticate > 407 for subscribe.What version of the SIP firmware are you using? I've had success with 1.3.0, 1.3.1, 1.3.4, and 1.4.1. My sip.conf entries for my Polycom phones look like this: [12] type=friend username=12 secret=12 callerid="12" host=dynamic dtmfmode=inband context=no-callwaiting mailbox=12@from-sip disallow=all allow=ulaw Are you configuring directly on the phone, or using an FTP or TFTP server?> Anybody could help me to configure Asterisk in order > to set instant message and presence ?To the best of my knowledge the Presence feature of the Polycom phones does not work with Asterisk. I believe it only works with other IM clients. Hope this helps! Noah
I'm able to register sip friends with asterisk but i wish to use presence and instant messaging. asterisk sent back "Proxy Authentication required 407" when SUBSCRIBE is sent to asterisk. harry --- Noah Miller <noah@rosecompanies.com> a ?crit?:> Hi Harry - > > > Can you get SUBSCRIBE registration ?? > > I don't know what SUBSCRIBE registration is, but > looking at your > sip.conf, there's a couple of things I would change: > > [100] > type=friend > username=100 > secret=100 > > fromuser=100 > ; Take this out - it's only needed when you want > certain types of sip > proxies are trying to > ; register to this peer - not normally needed for > asterisk. > > host=dynamic > context=sip > > dtmfmode=rfc2833 > ; I'd use inband here. I've tried rfc2833 here, > too, but it doesn't > seem to work as well as inband > > progressinband=no > ; You don't really need this, and I think it doesn't > make sense if > you're doing rfc2833 dtmfmode > > On the Polycom side, you should use the following > info: > > Address: 100 > Auth User ID: 100 > Auth Password: 100 > > Other than the SIP server address, these are the > only important numbers > on the Polycom. > > - Noah > > > > > > --- Noah Miller <noah@rosecompanies.com> a > ?crit?: > >> Hi Harry - > >> > >>> I try to set up two lines per ip 300 phone, > >>> registration is ok but i get Failure to > >> authenticate > >>> 407 for subscribe. > >> > >> What version of the SIP firmware are you using? > >> I've had success with > >> 1.3.0, 1.3.1, 1.3.4, and 1.4.1. > >> > >> My sip.conf entries for my Polycom phones look > like > >> this: > >> > >> [12] > >> type=friend > >> username=12 > >> secret=12 > >> callerid="12" > >> host=dynamic > >> dtmfmode=inband > >> context=no-callwaiting > >> mailbox=12@from-sip > >> disallow=all > >> allow=ulaw > >> > >> Are you configuring directly on the phone, or > using > >> an FTP or TFTP > >> server? > >> > >> > >>> Anybody could help me to configure Asterisk in > >> order > >>> to set instant message and presence ? > >> > >> To the best of my knowledge the Presence feature > of > >> the Polycom phones > >> does not work with Asterisk. I believe it only > >> works with other IM > >> clients. > >> > >> Hope this helps! > >> Noah > >> > >> > > > > > > > > > > > > > > D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace > de stockage pour > > vos mails ! > > Cr?ez votre Yahoo! Mail sur > http://fr.mail.yahoo.com/ > >D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
Hello, if somebody is interested in Europe for 2 polycom soundpoint ip 300 for testing with Asterisk contact me out of list . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com
Just so you know who you're dealing with: ---------- Forwarded message ---------- From: harry gaillac <gaillacharry@yahoo.fr> Date: Jun 24, 2005 7:58 PM Subject: Re: [Asterisk-Users] polycom soundpoint ip 300 To: Wilson Pickett i piss on you Wilson Pickett Harry from France> > if somebody is interested in Europe for 2 polycom > > soundpoint ip 300 for testing with Asterisk > contact me > > out of list . > > By all means.
Wilson Pickett, I posted this mail for people interested in polycom ip300 for asterisk. Harry --- Wilson Pickett <spamsucks2005@gmail.com> a ?crit :> > yes with anybody interesting in ip phones for > > Asterisk > > So you can send them a childish insult? I wrote you > off list as > requested, and you wrote back something a 10 year > old would say in > school. What is your point, Harry? > _______________________________________________ > 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>___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com