--- dkwok <[EMAIL PROTECTED]> wrote: > Just got GS 101 phone and plugged into the network.
Peoplehere complain about these phones but I don't seem to have a problem, well not after getting them set up correctly. I'm running with Software Version: Program--1.0.4.39 Bootloader--1.0.0.13 HTML--1.0.0.20 > > Got ip setup however, the following problems arise: > > 1. when dialing an extension, I cannot further send any key tone to > Asterisk. I'm using "SIP info" also with "payload type" set to "101" > 2. there is no sound coming from the other end. For some reason I found I had to place the disallow=...allow=... stuff under [gs] putting it in [General] didn't seem to do the trick. I also put "reinvite=no" in [gs] I once had sound going only one way due to t stupid error in my firewall config. I was purposfully droping packets and logging each one of them. Are you running firewall software on your * server? "ethereal" or other ethernet sniffing software is usfull to debug this kind of stuff > > I have a sip.conf setup for GS: > [General] > disallow=all > allow=ulaw > allow=alaw > > [gs] > canreinvite=no > dtmfmode=info > > In the GS101 setting > rtp port = 5004 > sip port = 5060 > dtmf = sip info > codec = pcmu > codec = pcma > > Any pointer of a sample of config file would be most appreciate. > > -- > David Kwok > > Iaxtel/FWD # 17001813482 ext 1002 > > ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s ===== Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ _______________________________________________ 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
