Hi, i have a 504G with a 500S and i'm trying to make prk work. these are examples of assigned values in 500s unit keys. fnc=prk;sub=call-park@mydomain.com;ext=call-park@mydomain.com;vid=3 fnc=sd+blf;sub=*32@mydomain.com;ext=*32@mydomain.com;vid=3 fnc=sd+blf;sub=*31@mydomain.com;ext=*31@mydomain.com;vid=3 the values below is what i get on my proxy. notice that the subscribe for the fnc=prk has the From and To headers equal and the sip uri . shouldn't the To header be "To : sip:call-park@mydomain.com" ? i'm using rfc3265_4235 as the server type Thanks SUBSCRIBE sip:call-park@mydomain.com from sip:504G-WITH-500S@mydomain.com to sip:504G-WITH-500S@mydomain.com SUBSCRIBE sip:*32@mydomain.com from sip:504G-WITH-500S@mydomain.com to sip:*32@mydomain.com SUBSCRIBE sip:*32@mydomain.com from sip:504G-WITH-500S@mydomain.com to sip:*32@mydomain.com
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Hi Dan, thanks for the reply. you are making some wrong assumptions on the usage pattern but in the end i got to know what i wanted, "not possible". this is very disappointed as yealink and polycom phones do have this option (set to 480). Thank you.
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Hi Dan, thanks for replying. the issue is that the phone is in a list of devices to ring and the returned code 486 makes other devices not ring. if the call is reject or temp_unavailable (480) it will work as expected. is there a way to configure spa5xx to send 480 or any of the other mentioned ? Thanks
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Hi, is there a way to change the code the phone sends on dnd ? i have a spa504g that sends 486 (busy) on dnd and provider tells that it should one of 400/401/402/403/603 CALL_REJECTED. Thanks
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Hi, 1) this doesn't work (MWI SUBSCRIBE not sent) xxxxx sip.domain.com 2) this works (MWI SUBSCRIBE is sent) xxxxx xxxxx@sip.domain.com
3) perfect world
FromUser|Custom FromUser => <User_ID_1_>@ ($USER@$PROXY?) xxxxx $PROXY
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Hi, there seems to be a bug in spa5xx 7.5.5 firmware related to srtp. scenario 2 SPA504G phones connected to freeswitch with s-descriptor on srtp method and secure call settings ON freeswitch by default will offer both RTP/SAVP and RTP/AVP when briding the call to phone B. phone A calls phone B with srtp enabled. freeswitch accepts the call and bridges the call to phone b. since phone A started the call with srtp, freeswitch passes to phone B the option to secure the call and sends both RTP/SAVP and RTP/AVP o=FreeSWITCH 1380857320 1380857321 IN IP4 192.168.16.128 s=FreeSWITCH c=IN IP4 192.168.16.128 t=0 0 m=audio 22742 RTP/SAVP 0 8 9 4 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:TT94irJQHRiPIKBxxjHiCSa15oyOhSm0Q+94UcpA a=ptime:30 a=sendrecv m=audio 22742 RTP/AVP 0 8 9 4 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=sendrecv when this happens , phone will freeze and will not accept any key input (menu, numbers, etc) or calls. the only way is to turn off/on there is a parameter in freeswitch that allows to offer only RTP/SAVP and in that case it works very well with both sides of call being secure. i think the phone B should check both RTP/SAVP RTP/AVP and choose accordinly to his setup because, we don't have a way to know in advance if the phone has srtp enabled and would like to offer both methods (this works with other ip phones and softphones) any comments ? cheers
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Hi, i'm trying to use cli to configure dhcp option 66 but i'm getting this error below. thanks switch702df6(config-dhcp)#ip dhcp pool network Voice switch702df6(config-dhcp)#option 66 ascii-string " http://provisioning.domain.tld/$MA.xml " % Wrong number of parameters or invalid range, size or characters entered
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Hi, i need to use separate vlan for voice/data and provide dhcp . upstream router is very basic, doesn't support vlan or dhcp options for provisioning.
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Hi, i'm trying to configure SF302-08P in layer 3 mode. ports 1 to 8 have ip phones and desktops, port GE1 in connected to a router (not cisco) that links to the internet. i created vlan 100 and assigned to ports 1 to 8 (untagged) i created vlan 200 and assigned to port ge1 (untagged) activated dhcp server , created pool with 192.168.26.0/24 gw 192.168.26.1 |router A| (192.168.16.0/24 gw 192.168.16.1) | | | GE1 (192.168.16.14) | SF302-08P | | | SPA504G+DESKTOP | SPA504G+DESKTOP 192.168.26.2 192.168.26.4 192.168.26.3 192.168.26.5 the problem that i'm facing is that SPA504G and desktops access the internet with their inter address 192.168.26.0/24 and not 192.168.16.14. is there a way to achieve what i'm looking ? how does routing work in this case ? shouldn't there be some kind of NAT ? thank you
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Hi, i can successfuly connect my spa504g to a kamailio server and get mwi notifications using UDP. when i use tls, i successfuly register in the the server and also subscribe to mwi events. i can make calls and srtp also works. the problem is with kamailio server sending mwi notifications to spa504g when i use tls. do i have to create a mini certificate and configure it in spa50g for tls mwi notifications to work ? i don't know if the problm is with the server or with misconfiguration in spa504g. the initial SUBSCRIBE event is successfulll and i can get a response (mwi light goes on if i have a vm). the problem seems to be when the connection is initiated by the server. any thoughts ? hints ? thank you
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