> > 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
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