On Oct 26, 2007, at 10:07 AM, Andrea Lanza wrote: > >> >> Hi Andrea, >> >> I have a feeling it is settings. Can you tell me more about >> what you had to take care about when getting the settings correct? >> >> What about my sip.conf entry? >> > ... > here is my sip.conf > > [5199999] > ;sip phone test wellgate wg3701b > username=5199999 > secret=5199999 > type=friend > qualify=yes > nat=never > host=dynamic > port=5060 > context=from-internal > canreinvite=no > dial=SIP/5199999 > callerid=fax prova wellgate 5199999<5199999> > > [5188888] > ;sip phone test ht 486 > username=5188888 > secret=5188888 > type=friend > qualify=yes > nat=never > host=dynamic > port=5060 > context=from-internal > canreinvite=no > dial=SIP/5188888 > callerid=fax prova ht486 5188888<5188888> > > [5177777] > ;sip phone test ht 386 > username=5177777 > secret=5177777 > type=friend > qualify=yes > nat=never > host=dynamic > port=5060 > context=from-internal > canreinvite=no > dial=SIP/5177777 > callerid=ata handytone 386 5177777<5177777> > > >> If i don't use the ATA for "a while", when i make a test call to >> the one of >> the lines i go right to voicemail since >callweaver gets a message >> that the >> line is busy as evidenced from the callweaver cli: > >> -- Called grandstream11 >> -- Got SIP response 486 "Busy" back from 192.168.1.227 >> -- SIP/grandstream11-2428 is busy > >> In this same state, if i pick up the handset on the analog phone >> on the ATA i >> hear a click but i get no dial tone. >> Curious is that if i put the handset down, then pick it up again i >> get a dial >> tone and if i then hang up the hand set i >> can call the phone on the ATA and it rings. > >> Have you seen any of these problems while you were working on >> getting your >> settings correct? > > It seems like the previous call wasn't hangup correctly.... Try > using a peer > definition like mine, then have a look at > your extensions.conf. I also have > dtmfmode=inband > in my general room in sip.conf > hope this helps, > > Andrea
Hi Andrea, I tried a lot of things to cure the problem. What i did to get the grandstream 386 ATA to work was as follows: 1. Set the grandstream to NOT register for both extensions 2. Set the sip.conf entry for the grandstream to use a host=IP address: [grandstream11] type=friend secret=grandstream11 context=from-internal-sip ; the internal context controls what we can do (or sip?) qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted ;host=dynamic ; This device registers with us ;defaultip=192.168.1.227 ; is fixed IP -- but not using host since callweaver complained host=192.168.1.227 port=5060 canreinvite=no ; don't allow RTP voice traffic to bypass CallWeaver dtmfmode=info ; or RFC2833 or inband for disallow=all allow=ulaw dial=SIP/grandstream11 Now there is always a dial tone when i pick up the analog phone attached to the ATA and when a call comes trough it rings the phone instead of going directly to voicemail. What didn't make a difference was moving to callweaver-1.1.99-RC.20071031. I guess as much as i have tried to understand the "host" sip setting (IP or dynamic), "defaultip" and "registering" in regards to callweaver, i still don't have a clear picture. Where is someone supposed to go to read about these and other settings other than the installed conf files? Peace, Dan _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
