dkwok wrote:> Just got GS 101 phone and plugged into the network. > > Got ip setup however, the following problems arise: > > 1. when dialing an extension, I cannot further send any key tone to > Asterisk. > 2. there is no sound coming from the other end. > > 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. >Here is what my sip.conf file looks like for a grandstream phone: [sringwald] disallow=all host=dynamic allow=ulaw type=friend username=sringwald secret=<SOME SECRET> callerid=Steve <777777> canreinvite=no reinvite=no insecure=yes nat=yes dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=777777 ; Mailbox for message waiting indicator
--- dkwok <dkwok@iware.com.au> 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 chrisalbertson90278@yahoo.com Cell: 310-990-7550 Office: 310-336-5189 Christopher.J.Albertson@aero.org KG6OMK __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/
Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040121/e6d1a04b/smime.bin
On Thu, 22 Jan 2004, dkwok wrote:> Just got GS 101 phone and plugged into the network. > > Got ip setup however, the following problems arise: > > 1. when dialing an extension, I cannot further send any key tone to > Asterisk. > 2. there is no sound coming from the other end. > [gs] > canreinvite=no > dtmfmode=infoTo solve 1., use dtmfmode = rfc2833 or just leave it empty.> In the GS101 setting > rtp port = 5004 > sip port = 5060 > dtmf = sip infousing "via RTP (RFC2833)" here works fine for me.> codec = pcmu > codec = pcma > > Any pointer of a sample of config file would be most appreciate.WRT the codecs, Setting all 6 choices in the grandstream web interface has helped for me, most of the time. One phone required several reset cycles before it would accept new settings, though. Another one only accepted the new settings after unplugging/replugging the power supply. This one also lost its settings during another power supply. I guess these "phones" are just a bit flakey WRT their settings... HTH, Siggi