> 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 > If assigning a fixed ip to a peer it works, and assigning a dynamic ip doesn't, you could check if when it is in dynamic mode it succesfully registers to the callweaver box. If it registers, which is the ip shown using sip show peers ? is it the correct one ? are these devices (ata and cw) onthe same net ? is there any firewall/router/nat between them ?
As far as I know, there shouldn't be any difference registering a dynamic peer or defining a fixed ip peer: if differences actually are, then probably are networking / firewall issues Regarding your question about documentation.... I think the answer is searching the net. you can also have a look here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf Andrea
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