Re: [Asterisk-Users] Budgettone 100 phone Configuration
On Wednesday 04 June 2003 03:08 pm, Stephen R. Besch wrote: > >Hi Just received the above phone > > > >Does anyone have sip.conf and extension.conf example for the SIP phone > >working with the FXS w100p and the FXO tdm400d > > > >any help would be appreciated > > I can't help with the FXS/FXO stuff but I can tell you what I've done > with the Budgetone 100. The network settings will depend on your local > configuration, so I'll leave most of those out of this discussion. On > the phone: + + + UPDATE; once I removed the 723 it worked again. Guess I don't need it as * is reading the gsm recording not the BT phone... __ Thanks for this, I got my BT102 up and running fine. It did not flash the display indicating VM so I added g723sf and now it notifies and I can pick up vm w message button. However, once * starts recording voicemail it hangs up. I assumed it has to do with the BT not having gsm so I added g723sf to voicemail.conf but it is not enough. Any pointer? Thanks! -- Steve _ This sig is pending approval by SCO's legal team. Licensing fee's expected to be "fair", i.e. $999. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
On Wednesday 04 June 2003 03:08 pm, Stephen R. Besch wrote: > >Hi Just received the above phone > > > >Does anyone have sip.conf and extension.conf example for the SIP phone > >working with the FXS w100p and the FXO tdm400d > > > >any help would be appreciated > > I can't help with the FXS/FXO stuff but I can tell you what I've done > with the Budgetone 100. The network settings will depend on your local > configuration, so I'll leave most of those out of this discussion. On > the phone: Thanks for this, I got my BT102 up and running fine. It did not flash the display indicating VM so I added g723sf and now it notifies and I can pick up vm w message button. However, once * starts recording voicemail it hangs up. I assumed it has to do with the BT not having gsm so I added g723sf to voicemail.conf but it is not enough. Any pointer? Thanks! -- Steve _ This sig is pending approval by SCO's legal team. Licensing fee's expected to be "fair", i.e. $999. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
The latest release 1.0.3.60 will allow you to setup a outbound proxy for the phone and the SIP URI's domain can be set in SIP SERVER field. Any sip messages from the phone will use SIP SERVER field's content as domain for phone's SIP URI. --- Juha Heinanen <[EMAIL PROTECTED]> wrote: > Dan Fernandez writes: > > > In the phone, if I set the outbound proxy to the linksys it > doesn磘 do > > anything. > > i have noticed this too. outbound proxy feature is broken in it. > also, > it doesn't do srv lookups, which would allow leaving outbound proxy > empty. it looks to me that the gs guys still have ways to go before > the > phone is ready for prime time. > > -- juha > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
If your proxy server has the same DNS domain name as your SIP URI, then leave outbound proxy blank. Then, check "yes" for NAT Traversal. Type in the STUN server ip, GS has a STUN server 67.153.142.67, or you can use vovida's 66.7.238.210. BTW, there is a new release, check the web site: www.grandstream.com/y-service.htm. You can configure your GS phone with GS tftp server: 4.3.153.56, the reboot the phone. Since you use 1.0.3.53, you will lose you phone configuration this time, you will not lose any confituration from 1.0.3.58 and later release. --- Dan Fernandez <[EMAIL PROTECTED]> wrote: > Will look into this once someone can help me with the configuration > behind > NAT (without NAT I have no problem) > I am using v1.0.3.53 and a linksys router (the phone IP is > 192.168.1.2) > > I磛e tried in my sip.conf with and without NAT=1. > > In the phone, if I set the outbound proxy to the linksys it doesn磘 > do > anything. If I leave outbound proxy empty it registers and I can > place calls > but no audio either way. I have also tried setting the phone for NAT > and no > NAT (no STUN server). > > Don磘 know what else to try. Can someone please help me? > > > - Original Message - > From: "Greg Renouf" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, June 05, 2003 8:16 PM > Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration > > > > I'm using v.1.0.3.58 and am experiencing that my phone crashes > every > > time the call reaches about 45 minutes in length. > > > > Has anybody had a similar experience? > > > > -GSR > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Stephen > R. > > Besch > > Sent: 05 June 2003 19:03 > > To: [EMAIL PROTECTED] > > Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration > > > > The updated Budgetone firmware (1.0.3.60) has indeed fixed the > "silent > > DTMF" issue. > > > > >By the way, Grandstream just got the "silent DTMF" problem fixed > for > > me > > >and sent me an updated revision this morning (1.0.3.60). I am > just > > >about to install it, but it may require that I debug my tftp > server, > > >which I haven't tested yet. I'll post the list as soon as I get > the > > >new version loaded. > > -- > > Stephen R. Besch, Ph.D. > > SachsLab > > 320 Cary Hall > > SUNY at Buffalo > > Buffalo, NY 14214 > > (716) 829-3289 x106 > > > > ___ > > 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 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
I've never seen a Linksys router that was a proxy. They are all NAT routers. Dan Fernandez writes: In the phone, if I set the outbound proxy to the linksys it doesn´t do anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Dan Fernandez writes: > In the phone, if I set the outbound proxy to the linksys it doesn´t do > anything. i have noticed this too. outbound proxy feature is broken in it. also, it doesn't do srv lookups, which would allow leaving outbound proxy empty. it looks to me that the gs guys still have ways to go before the phone is ready for prime time. -- juha ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Will look into this once someone can help me with the configuration behind NAT (without NAT I have no problem) I am using v1.0.3.53 and a linksys router (the phone IP is 192.168.1.2) I´ve tried in my sip.conf with and without NAT=1. In the phone, if I set the outbound proxy to the linksys it doesn´t do anything. If I leave outbound proxy empty it registers and I can place calls but no audio either way. I have also tried setting the phone for NAT and no NAT (no STUN server). Don´t know what else to try. Can someone please help me? - Original Message - From: "Greg Renouf" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 05, 2003 8:16 PM Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration > I'm using v.1.0.3.58 and am experiencing that my phone crashes every > time the call reaches about 45 minutes in length. > > Has anybody had a similar experience? > > -GSR > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. > Besch > Sent: 05 June 2003 19:03 > To: [EMAIL PROTECTED] > Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration > > The updated Budgetone firmware (1.0.3.60) has indeed fixed the "silent > DTMF" issue. > > >By the way, Grandstream just got the "silent DTMF" problem fixed for > me > >and sent me an updated revision this morning (1.0.3.60). I am just > >about to install it, but it may require that I debug my tftp server, > >which I haven't tested yet. I'll post the list as soon as I get the > >new version loaded. > -- > Stephen R. Besch, Ph.D. > SachsLab > 320 Cary Hall > SUNY at Buffalo > Buffalo, NY 14214 > (716) 829-3289 x106 > > ___ > 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 > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgettone 100 phone Configuration
I'm using v.1.0.3.58 and am experiencing that my phone crashes every time the call reaches about 45 minutes in length. Has anybody had a similar experience? -GSR -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: 05 June 2003 19:03 To: [EMAIL PROTECTED] Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration The updated Budgetone firmware (1.0.3.60) has indeed fixed the "silent DTMF" issue. >By the way, Grandstream just got the "silent DTMF" problem fixed for me >and sent me an updated revision this morning (1.0.3.60). I am just >about to install it, but it may require that I debug my tftp server, >which I haven't tested yet. I'll post the list as soon as I get the >new version loaded. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ 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
re: [Asterisk-Users] Budgettone 100 phone Configuration
The updated Budgetone firmware (1.0.3.60) has indeed fixed the "silent DTMF" issue. >By the way, Grandstream just got the "silent DTMF" problem fixed for me >and sent me an updated revision this morning (1.0.3.60). I am just >about to install it, but it may require that I debug my tftp server, >which I haven't tested yet. I'll post the list as soon as I get the >new version loaded. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgettone 100 phone Configuration
>>mailbox=100 ;Set to use MWI on phone >Stephen, does your MWI work? It doesn't seem to work for me - I assumed >it had not been implemented yet. >Phil. Phil, Yes it does. My firmware revision is 1.0.3.58. But keep in mind, that, unfortunately, the only MWI indication is a blinking of the LCD and a stutter dialtone. As I recall, the display also blinks during boot and during some error conditions, but this should not really be a problem once the phones are set up and stable.. By the way, Grandstream just got the "silent DTMF" problem fixed for me and sent me an updated revision this morning (1.0.3.60). I am just about to install it, but it may require that I debug my tftp server, which I haven't tested yet. I'll post the list as soon as I get the new version loaded. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgettone 100 phone Configuration
-Original Message- From: Stephen R. Besch [mailto:[EMAIL PROTECTED] Sent: 04 June 2003 20:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Budgettone 100 phone Configuration > mailbox=100 ;Set to use MWI on phone Stephen, does your MWI work? It doesn't seem to work for me - I assumed it had not been implemented yet. Phil. ___ 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
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Thanks for every ones help so far I can now dial the sip phone and it
rings and I can talk
But I cannot dial another extesion from it I just get a 404 error , but
in sip debug the phone is calling 699
Sip read:
ACK sip:@10.1.1.76 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.77
From: "robbsip"
;tag=6021754f-b8fc-4ff0-6d90-932f03045026
To: ;tag=as5b106bd1
Contact:
Proxy-Authorization: DIGEST username="robb", realm="asterisk",
algorithm=MD5, uri="sip:@10.1.1.76", nonce="3f6cebfc",
response="5320b41c1333655f34889a104016f3e9"
Call-ID: [EMAIL PROTECTED]
CSeq: 18159 ACK
User-Agent: Grandstream SIP UA 1.0.3.56
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Length: 0
here is my extension .conf
[incoming]
exten => s,1,Dial,Zap/2|20
;exten => s,1,Dial,sip/robb
exten => s,2,Voicemail,u1000
exten => s,102,Voicemail,b1000
[outgoing]
exten => 1234,1,VoiceMailMain
exten => _9,1,Dial,Zap/1/${EXTEN}
exten => _9XXX,1,Dial,Zap/1/${EXTEN}
exten => _22X,1,Dial,Zap/1/${EXTEN}
;zap ch 2
exten => 699,1,Dial,Zap/2-1
; Extension 698 - Grandstream
exten => 698,1,Dial,sip/robb|10|t; Ring, 10 secs max
; Ext 697 martin
exten => 697,1,Dial,sip/martin|10|t
[sip]
any more help would be appreciated
Robb
Skuse, Phil wrote:
Here's the config I am using for my budgie 100, although I don't use FXS or
FXO cards.
FYI Grandstream posted a firmware update on their website a couple of days
ago.
sip.conf
[698] ; Grandstream Phone
type=friend
insecure=yes
host=dynamic
permit=172.16.0.0/255.255.0.0
mailbox=698
dtmfmode=inband
extensions.conf
; Extension 698 - Grandstream
exten => 698,1,Playback,transfer|skip ; "Please hold while..."
exten => 698,2,Dial,sip/698|10|t; Ring, 10 secs max
exten => 698,3,Voicemail,u698 ; Send to voicemail...
exten => 698,5,Goto,s|6 ; Start over
exten => 698,103,Voicemail,b698 ; (2 + 101) "I'm on the phone"
exten => 698,104,Goto,5 ; Go to voicemail, etc.
-----Original Message-----
From: Robert Boardman [mailto:[EMAIL PROTECTED]
Sent: 04 June 2003 12:49
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Budgettone 100 phone Configuration
Hi Just recieved the above phone
Does anyone have sip.conf and extension.conf example for the SIP phone
working
with the FXS w100p and the FXO tdm400d
any help would be appreciated
Thanks
Robb
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[Asterisk-Users] Budgettone 100 phone Configuration
>Hi Just received the above phone
>Does anyone have sip.conf and extension.conf example for the SIP phone
>working with the FXS w100p and the FXO tdm400d
>any help would be appreciated
I can't help with the FXS/FXO stuff but I can tell you what I've done
with the Budgetone 100. The network settings will depend on your local
configuration, so I'll leave most of those out of this discussion. On
the phone:
1) Set the phone up to use in-band signaling. At the present time,
out-of-band does not work reliably. It has something to do with the
fact that the RFC2833 specifies that repeated RTP packets can be sent as
long as a button is pressed. Asterisk sees these as multiple digits.
Budgetone is working on this issue, and I believe so is Mark Spencer.
2) I have found that dynamic registration only works if the SIP User ID
and the Authenticate ID on the phone both match the section title for
the phone in sip.conf. It doesn't matter if you statically assign the
IP or use DHCP. (see note below)
3) Set "Early Dial" to "yes"
4) Set "Use # as dial key" to "no"
5) If you are using the voicemail application, you can set the
"voicemail user ID" to automatically open your voice mail box. You will
need to create a unique extension and then enter it in the voicemail
user ID field with your mail box appended. For example, assuming that
you are using 3 digit mailboxes and you choose to define your mailbox
retrieval extension as _78XXX, then put 78100 in this field. Then to
get your messages, pick up the handset and press the message button.
6) The budgetone phone needs access to an NTP server (at least for now)
to set the Date/Time. If you are running your phones on a non-routable
network, then you will need to mirror an NTP server through your
asterisk server on the same subnet. I did this by adding the following
line to \etc\ntp.conf:
restrict 192.168.10.0 mask 255.255.255.0 notrust nomodify notrap
This permits any phone with an IP address from 192.168.10.0 to
192.168.10.255 to get the date and time from your linux box. Just
change the IP base address and netmask to the range you want to use and
insert this line in the conf file.
7) If you want to update the phone's firmware, you will need access to a
tftp server. You can enable your own, or use the one that grandstream
provides (see their web site)
8) There are some issues with the sounding of the DTMF tones. Under some
circumstances, when you press a button, there will be no sound. The
tone packets are sent to asterisk, just no sound is heard. Budgetone is
working on this issue and it should be fixed very soon. Check their web
site for updated firmware in the next week or so.
9) If you specify a mailbox in the phones definition in sip.conf, any
time there are unheard messages in that inbox, SIP MWI packets will be
sent to the phone. The phone will blink the display and deliver a
stutter dial tone if there are messages waiting. When you empty the
inbox, the display stops flashing and the stutter dial tone is replaced
with a standard dial tone.
--
Then, in sip.conf, add an entry for each phone. For example, for an
extension numbered 100 with a voicemail box defined as 100
[budgetone100] ;I name each phone as type + exten
type=friend
context=longdistance;or some other appropriate context
username=yourname
fromuser=Your Full Name
host=Phones IP address ; or dynamic
dtmfmode=inband ;important! rfc2833 may work in future
secret=@@##!00 ;Optional, only works if dynamic
qualify=1000;If set, asterisk will test line response time
mailbox=100 ;Set to use MWI on phone
In extensions.conf. I put this in my "local" context so that people
calling in from outside could not access the "automatic" message
retrieval extension. There is different extension in the default
context for accessing voice mail from the outside, which requires the
entry of the mailbox and password. I also bypass the password check in
our system, since everyone has their own phone and message security is
not an issue for us. Remove the "s" from the VoicemailMain argument if
you want to enforce password usage.
exten => _781XX,1,Wait(1)
exten => _781XX,2,VoicemailMain(s${EXTEN:2})
exten => _781XX,3,Hangup
In the default context (assumes that you are using a stdexten macro):
exten => 107,1,Macro(stdexten,SIP/budgetone100)
Extras:
These are my notes on the "host=" option. Some of it was gleaned by
studying the sources, some by trial and error. If the experts would
critique and correct it would be appreciated.
option: host=
Valid in:
sip.conf
Others???
Format:
host= - or-
host=dynamic
Function:
Defines the IP address of a SIP (or other) type phone.
Using host=dynamic. This option is not quite what it appears to be. The
idea behind it is that it permits the phone to define
RE: [Asterisk-Users] Budgettone 100 phone Configuration
Here's the config I am using for my budgie 100, although I don't use FXS or FXO cards. FYI Grandstream posted a firmware update on their website a couple of days ago. sip.conf [698] ; Grandstream Phone type=friend insecure=yes host=dynamic permit=172.16.0.0/255.255.0.0 mailbox=698 dtmfmode=inband extensions.conf ; Extension 698 - Grandstream exten => 698,1,Playback,transfer|skip ; "Please hold while..." exten => 698,2,Dial,sip/698|10|t; Ring, 10 secs max exten => 698,3,Voicemail,u698 ; Send to voicemail... exten => 698,5,Goto,s|6 ; Start over exten => 698,103,Voicemail,b698 ; (2 + 101) "I'm on the phone" exten => 698,104,Goto,5 ; Go to voicemail, etc. -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 04 June 2003 12:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Budgettone 100 phone Configuration Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb ___ 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
[Asterisk-Users] Budgettone 100 phone Configuration
Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
