Key Aavoja
2004-Jan-23 03:00 UTC
[Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100
Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to register in asterisk if Im trying to call somewhere. I searched in list-archive and I didnt found that anybody else has this kind of problem. I read also: http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html and I did so. sip.conf --------- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all ; Disallow all codecs allow=g729 [cisco] context=in type=friend insecure=yes host=<removed> dtmfmode=rfc2833 [grandstream1] type=friend secret=grandstream1 host=dynamic context=class1 dtmfmode=rfc2833 [grandstream2] type=friend secret=grandstream2 nat=yes host=dynamic context=class1 dtmfmode=rfc2833 Asterisk ver: Asterisk CVS-01/22/04-18:13:23 Grandstream ver: Program--1.0.3.81 Bootloader--1.0.0.7 HTML--1.0.0.18 * And as I mentioned before, without registration and with static IP everything works, it seems, that something is misconfigured in my setup for authentication or this phone firmware is buggy? (but its latest, I checked www.grandstream.com) ____________________________ Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/
Larry Keyes
2004-Jan-23 07:41 UTC
[Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100
Key, I've been playing with the Grandstreams for some weeks; one good way to see the registration messages it to monitor the network with Ethereal. (packet sniffer). You'll see the SIP messages coming and going, with complete decoding. This works pretty much as predicted when using VOCAL. (another SIP server) That said, I recently asked about registration with Asterisk from the Grandstreams, and received a reply that if the phones have a fixed IP address they do not "register" as they are already accounted for in the sip.conf by the user name/number assigned to the phone, and indeed the phones work fine (at least inside the firewall). Other soureces have described Asterisk as having "limited or not fully implemented SIP support"... but I have never been able to determine to what extent SIP support is lacking. I was able to get the Grandstreams running only recently with Asterisk... My sip.conf for the Grandstreams has several differences from yours: For a grandstream phone with an assigned number of 1000 and a fixed IP address of 192.168.0.160 I have the following: [1000] type=friend username=1000 host=dynamic reinvite=no canreinvite=no qualify=300 callerid="Larry's Desk" <1000> mailbox=1000 nat=no dtmfmode=info disallow=all allow=ulaw allow=mlaw See the message I sent to Tom Scott on 1/21 for a little more information. I would very much like to put together a set of instructions for the Grandstream phones+asterisk that could expand on the scattered information found on the wiki/web/draft manual and this list. And let us know what you find! -- L
Stephen R. Besch
2004-Jan-23 09:47 UTC
[Asterisk-Users] Re: SIP register/auth with Grandstream BudgeTone-100
Key Aavoja wrote:> Hello, > > I have a problem with asterisk and Grandstream BudgeTone-100. > With default configuration everything works (in anonymous mode and fixed > IP), but if Im trying to enable registering, it dos not work. > I used 'sip debug' and verbose level 10, nothing happens if I switch > telephone on (no messages about bad auth etc). As I understood, after > switching phone on at first it will try to register in asteriskYes if Im> trying to call somewhere.Registers before any calls are made. Probably your extension name and registration data don't match. Here is my SIP config and a list of the GS phone settings: [exten106] type=friend context=administrator callerid=<829-3289 106> username=sbesch host=dynamic dtmfmode=info ;or inband if you prefer secret=yourpassword qualify=5000 mailbox=106 canreinvite=no ;As long as the phones are NAT'ed The caller ID only means something to our internal extensions, since the phone company will not let me set callerid data. I don't think that the username is needed. I use it because it shows up in the CLI response to SIP SHOW PEERS and helps me identify the phone. The important bits are that the extension name (the part in "[]") and the secret match the data in the phone setup: Sip User ID: exten106 Authenticate ID: exten106 Aithentication Password: yourpassword Sip Registration: Yes Send DTMF: via SIP Info Don't make the mistake of thinking that the username entry in SIP.conf has anything to do with the Authenticate ID. IT doesn't. The only thing that works is to set the User ID and the Authenticate ID to the same thing. You may find that the GS phones will dissappear after a while if you use dynamic registration. Alas, this is a bug in the GS firmware(1.0.3.81). I don't know if it has been fixed in later releases. I am not willing to update my phones until the firmware gets much more stable - they are all working and my philosophy is that if it ain't broke, don't fix it - so I haven't been able to test this. If your phones have fixed addresses, you might as well specify the IP addresses in SIP.conf and preemptively avoid the problem of the GS registrations dissappearing.
bfracall@hotmail.com
2004-Jan-26 00:07 UTC
[Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100
Hi,Key Aavoja, Have you successfully registed to * with secret specificated? Regards. bfrac ----- Original Message ----- From: "Key Aavoja" <key@neoon.com> To: <asterisk-users@lists.digium.com> Sent: Friday, January 23, 2004 2:00 AM Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100> Hello, > > I have a problem with asterisk and Grandstream BudgeTone-100. > With default configuration everything works (in anonymous mode and fixed > IP), but if Im trying to enable registering, it dos not work. > I used 'sip debug' and verbose level 10, nothing happens if I switch > telephone on (no messages about bad auth etc). As I understood, after > switching phone on at first it will try to register in asterisk if Im > trying to call somewhere. > > I searched in list-archive and I didnt found that anybody else has this > kind of problem. I read also: > http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html > and I did so. > > sip.conf > --------- > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = default ; Default for incoming calls > disallow=all ; Disallow all codecs > allow=g729 > > [cisco] > context=in > type=friend > insecure=yes > host=<removed> > dtmfmode=rfc2833 > > [grandstream1] > type=friend > secret=grandstream1 > host=dynamic > context=class1 > dtmfmode=rfc2833 > > [grandstream2] > type=friend > secret=grandstream2 > nat=yes > host=dynamic > context=class1 > dtmfmode=rfc2833 > > Asterisk ver: Asterisk CVS-01/22/04-18:13:23 > > Grandstream ver: Program--1.0.3.81 Bootloader--1.0.0.7HTML--1.0.0.18> > * And as I mentioned before, without registration and with static IP > everything works, it seems, that something is misconfigured in my setup > for authentication or this phone firmware is buggy? (but its latest, I > checked www.grandstream.com) > > > > ____________________________ > Best Regards: > Key Aavoja > > > > > /* Never argue with an idiot. They drag you down to their level, then beat > you with experience.*/ > > _______________________________________________ > 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 > >