>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.
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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,HangupIn 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=<ip address> - 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 the IP address rather than having it defined in sip.conf. It doesn't matter how the IP is obtained or set up at the phone. If the IP is specified in sip.conf with "host=", asterisk will attempt to communicate with the phone when the conf file is loaded, and from time to time thereafter, in an attempt to establish a registration. In this case, the "secret=" option appears to have no effect. If the host is specified as "dynamic", asterisk will do nothing until the phone sends an SIP:REGISTER request. This request includes the password and username specified by the phone. Asterisk first checks the username against each of the "[sipphone]" entries defined in sip.conf until it finds a match. If no match is found, the registration request is rejected. If there is a "secret=" option, asterisk next checks the password to see if it matches the "secret=" field. If it doesn't, the request is rejected. Once the password is validated, the phone will be registered and communication may begin. As far as I can tell, the name must match the section title (the name in the matching "[]". Matching the "username=field" doesn't work.
Note that there is really no direct relationship between the host=dynamic" option and DHCP, although it does allow interoperability with systems that use DHCP, given that, in general, the IP isn't known
when asterisk starts. All dynamic means is that the phone supplies the IP, making management of sip.conf easier and also permitting authentication.
-- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106
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