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snom 305 blf visual alert in display

hi there.


we have a freepbx system running and i am evaluating if snom phones can be a replacement for our current lineup of old phones.


this is about the blf capabilities of the firmware. blf and the sip subscription is working fine, the blf assigned button's led will blink when a call hits the monitored extension, and call pickup works too.


what i would like to accomplish is some kind of visual blf alert in the display. eg have the display prompt the calling number, so that a user can decide if he wants to pick the call or not.


i know sangoma phones can do that. can snom phones too? how would i proceed to do so? does it depend on the model i am using, which is currently a basic model 305.


thank you for any help or ideas.


best,

sebastian


Best Answer

to see how things go on a default install, i started from scratch on fresh installation of freepbx 14.0.3.1 with asterisk 15.4.0.


to finalize my individual setup i changed this:

- set chan_sip to udp 5060 (i want to use chan_sip and not pjsip)

- add fiield "notifycid = yes" in chan sip settings in other sip settings.


after that i created three extensions.


on the snom phone d375 i set a function key to "blf": context to ext@freepbx_ip (and not "active"), and number to "sip:other_ext@freepbx_ip|**". the string will be constructed by the phone, i only typed into the field "other_ext|**". where freepbx_ip is something like 10.10.10.1 and other_ext is something like 1234.


and voila, everythin is running so smoothly. blf with call screen almost out of the box. very nice.


so what was different now? at first it is the contexts. in the setup above, all contexts are set to the freepbx defaults. playing around with contexts you really need to know where the calls will go and what happens behind the scenes.


second, it is the asterisk version. with the default install using freepbx 14 with asterisk 13 it did not work at all!


i hope that helps anyone with the same requirement on snom phones.


best,

sebastian



Hi Sebastian, 


if I understood what do you want to achieve I think you are looking for the following setting


Best regard

Snom Support Team

hi sidaty.


thank you for your reply.


i guess the documention (your link) describes the feature i am after, but i still cannot get it to work. i have a 305 and a 375, both show the same behaviour: blf led is blinking, pickup does work, but nothing shows up in the display. goto_monitor_state_on_line_activity is set to "on".


from the documentation: "When any of your monitored lines shows an activity (other than idle), the phone will automatically display the call-monitor state."


but it does not, the display does not change at all. in asterisk and in the phones webinterface, i do see the active subscriptions.


is there some other parameter which needs to be set? how does the call-monitor state look like?


thank you for your help,

sebastian

hi Sebastian, 


there is another setting that could affect this behaviour. 

See more under: 

http://wiki.snom.com/Settings/pui_states_allowing_state_switch_on_activity


how does the call-monitor state look like?

see attached pic!


Best regards

Snom Support Team

bmp

hi sidathy.


thanks for getting back to me.


so your screenshot exactly shows what i want to accomplish.


the xml settings you refer to is already "on" with valid states being "idle". that makes sense to me, since the phone might only show the calls screen while i am not in a call.


i am a bit out of ideas now. does the call screen depend on any further asterisk setup? just as an additional information the subscription type is shown as:

Connected to Asterisk 1.8.32.2 currently running on freepbx (pid = 5033)
Verbosity is at least 3
freepbx*CLI> sip show subscriptions
Peer             User             Call ID          Extension        Last state     Type            Mailbox    Expiry
10.254.102.114   1113             313532373531353  1114@ext-local   Idle           dialog-info+xml <none>     003600
1 active SIP subscription

where 1113 is my extension, and 1114 the extension i am monitoring.


i have a feeling, the subscription lacks something that is required for the phone to accept the notify packet as a trigger. i just traced the notify from asterisk to my phone, and i found that no caller-id is transmitted. my idea of the calls screen is that you need both the called id and the calling id to display a reasonable information.


thanks for your help. i really would like to get this going, because the phones really make a good impression to me.


thank you,

best,

sebastian


to provide more input, this is a freepbx sip set debug on trace.


  

<------------>
Scheduling destruction of SIP dialog '0fd55323ddc79fa08c342ff893dec365000k4g0@provider_net.136.99' in 32000 ms (Method: OPTIONS)

<--- SIP read from TCP:provider_net.136.115:49998 --->
INVITE sip:+496955551114@our_ip:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP provider_net.136.115:5060;branch=z9hG4bKa6k11810bobmaofg54c0.1
Call-ID: SDo0f9801-dbeec90b68019b72a92491be51913d95-l65h813
From: <sip:+4917844444188@ip_sip_provider;user=phone>;tag=SDo0f9801-l9lesar1-CC-37
To: <sip:+496955551114@ip_sip_provider;user=phone>
CSeq: 1 INVITE
Max-Forwards: 61
Contact: <sip:+4917844444188@provider_net.136.115:5060;transport=tcp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Content-Length: 348
Content-Type: application/sdp
P-QSC-Relay: d=softx2tnb
X-CID: a9a88ers809e98nr15e10208hl9s5e59@SoftX3000

v=0
o=HuaweiSoftX3000 15695908 15695908 IN IP4 provider_net.136.115
s=Sip Call
c=IN IP4 provider_net.136.115
t=0 0
m=audio 23580 RTP/AVP 8 0 18 4 2 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
<------------->
--- (14 headers 15 lines) ---
Sending to provider_net.136.115:5060 (no NAT)
Using INVITE request as basis request - SDo0f9801-dbeec90b68019b72a92491be51913d95-l65h813
Found peer 'SIP_TRUNK' for '+4917844444188' from provider_net.136.115:49998
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port provider_net.136.115:23580
Looking for +496955551114 in from-trunk-sip-SIP_TRUNK (domain our_ip)
list_route: hop: <sip:+4917844444188@provider_net.136.115:5060;transport=tcp>

<--- Transmitting (no NAT) to provider_net.136.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP provider_net.136.115:5060;branch=z9hG4bKa6k11810bobmaofg54c0.1;received=provider_net.136.115
From: <sip:+4917844444188@ip_sip_provider;user=phone>;tag=SDo0f9801-l9lesar1-CC-37
To: <sip:+496955551114@ip_sip_provider;user=phone>
Call-ID: SDo0f9801-dbeec90b68019b72a92491be51913d95-l65h813
CSeq: 1 INVITE
Server: FPBX-2.11.0(1.8.32.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+496955551114@our_ip:5060;transport=TCP>
Content-Length: 0


<------------>
    -- Executing [+496955551114@from-trunk-sip-SIP_TRUNK:1] Set("SIP/SIP_TRUNK-000076f8", "GROUP()=OUT_2") in new stack
    -- Executing [+496955551114@from-trunk-sip-SIP_TRUNK:2] Goto("SIP/SIP_TRUNK-000076f8", "from-trunk,+496955551114,1") in new stack
    -- Goto (from-trunk,+496955551114,1)
    -- Executing [+496955551114@from-trunk:1] Set("SIP/SIP_TRUNK-000076f8", "__FROM_DID=+496955551114") in new stack
    -- Executing [+496955551114@from-trunk:2] Gosub("SIP/SIP_TRUNK-000076f8", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/SIP_TRUNK-000076f8", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/SIP_TRUNK-000076f8", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/SIP_TRUNK-000076f8", "") in new stack
    -- Executing [+496955551114@from-trunk:3] Set("SIP/SIP_TRUNK-000076f8", "CDR(did)=+496955551114") in new stack
    -- Executing [+496955551114@from-trunk:4] ExecIf("SIP/SIP_TRUNK-000076f8", "1 ?Set(CALLERID(name)=+4917844444188)") in new stack
    -- Executing [+496955551114@from-trunk:5] Set("SIP/SIP_TRUNK-000076f8", "CHANNEL(musicclass)=default") in new stack
    -- Executing [+496955551114@from-trunk:6] Set("SIP/SIP_TRUNK-000076f8", "__MOHCLASS=default") in new stack
    -- Executing [+496955551114@from-trunk:7] Set("SIP/SIP_TRUNK-000076f8", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [+496955551114@from-trunk:8] Set("SIP/SIP_TRUNK-000076f8", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [+496955551114@from-trunk:9] Goto("SIP/SIP_TRUNK-000076f8", "from-did-direct,1114,1") in new stack
    -- Goto (from-did-direct,1114,1)
    -- Executing [1114@from-did-direct:1] Set("SIP/SIP_TRUNK-000076f8", "__RINGTIMER=18") in new stack
    -- Executing [1114@from-did-direct:2] Macro("SIP/SIP_TRUNK-000076f8", "exten-vm,novm,1114,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/SIP_TRUNK-000076f8", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/SIP_TRUNK-000076f8", "TOUCH_MONITOR=1527518478.37541") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/SIP_TRUNK-000076f8", "AMPUSER=+4917844444188") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/SIP_TRUNK-000076f8", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/SIP_TRUNK-000076f8", "1?Set(REALCALLERIDNUM=+4917844444188)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/SIP_TRUNK-000076f8", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/SIP_TRUNK-000076f8", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/SIP_TRUNK-000076f8", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/SIP_TRUNK-000076f8", "1?report") in new stack
    -- Goto (macro-user-callerid,s,16)
    -- Executing [s@macro-user-callerid:16] GotoIf("SIP/SIP_TRUNK-000076f8", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:17] Set("SIP/SIP_TRUNK-000076f8", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:18] GotoIf("SIP/SIP_TRUNK-000076f8", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] Set("SIP/SIP_TRUNK-000076f8", "CALLERID(number)=+4917844444188") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/SIP_TRUNK-000076f8", "CALLERID(name)=+4917844444188") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/SIP_TRUNK-000076f8", "CDR(cnum)=+4917844444188") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/SIP_TRUNK-000076f8", "CDR(cnam)=+4917844444188") in new stack
    -- Executing [s@macro-user-callerid:33] Set("SIP/SIP_TRUNK-000076f8", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/SIP_TRUNK-000076f8", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/SIP_TRUNK-000076f8", "__EXTTOCALL=1114") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/SIP_TRUNK-000076f8", "__PICKUPMARK=1114") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/SIP_TRUNK-000076f8", "RT=") in new stack
    -- Executing [s@macro-exten-vm:6] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [s@macro-exten-vm:7] ExecIf("SIP/SIP_TRUNK-000076f8", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:8] Gosub("SIP/SIP_TRUNK-000076f8", "sub-record-check,s,1(exten,1114,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/SIP_TRUNK-000076f8", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/SIP_TRUNK-000076f8", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("SIP/SIP_TRUNK-000076f8", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/SIP_TRUNK-000076f8", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("SIP/SIP_TRUNK-000076f8", "0?exten,1") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/SIP_TRUNK-000076f8", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/SIP_TRUNK-000076f8", "NOW=1527518478") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/SIP_TRUNK-000076f8", "__DAY=28") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/SIP_TRUNK-000076f8", "__MONTH=05") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/SIP_TRUNK-000076f8", "__YEAR=2018") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/SIP_TRUNK-000076f8", "__TIMESTR=20180528-164118") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/SIP_TRUNK-000076f8", "__FROMEXTEN=+4917844444188") in new stack
    -- Executing [s@sub-record-check:21] Set("SIP/SIP_TRUNK-000076f8", "__CALLFILENAME=exten-1114-+4917844444188-20180528-164118-1527518478.37541") in new stack
    -- Executing [s@sub-record-check:22] Goto("SIP/SIP_TRUNK-000076f8", "exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] GotoIf("SIP/SIP_TRUNK-000076f8", "0?callee") in new stack
    -- Executing [exten@sub-record-check:2] Set("SIP/SIP_TRUNK-000076f8", "__REC_POLICY_MODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:3] GotoIf("SIP/SIP_TRUNK-000076f8", "1?caller") in new stack
    -- Goto (sub-record-check,exten,10)
    -- Executing [exten@sub-record-check:10] Set("SIP/SIP_TRUNK-000076f8", "__REC_POLICY_MODE=") in new stack
    -- Executing [exten@sub-record-check:11] GosubIf("SIP/SIP_TRUNK-000076f8", "0?record,1(exten,1114,+4917844444188)") in new stack
    -- Executing [exten@sub-record-check:12] Return("SIP/SIP_TRUNK-000076f8", "") in new stack
    -- Executing [s@macro-exten-vm:9] GotoIf("SIP/SIP_TRUNK-000076f8", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,15)
    -- Executing [s@macro-exten-vm:15] GosubIf("SIP/SIP_TRUNK-000076f8", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:16] Macro("SIP/SIP_TRUNK-000076f8", "dial-one,,Ttr,1114") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/SIP_TRUNK-000076f8", "DEXTEN=1114") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/SIP_TRUNK-000076f8", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/SIP_TRUNK-000076f8", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/SIP_TRUNK-000076f8", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/SIP_TRUNK-000076f8", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/SIP_TRUNK-000076f8", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/SIP_TRUNK-000076f8", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("SIP/SIP_TRUNK-000076f8", "EXTHASCW=") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/SIP_TRUNK-000076f8", "1?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,12)
    -- Executing [s@macro-dial-one:12] GotoIf("SIP/SIP_TRUNK-000076f8", "0?docfu:skip3") in new stack
    -- Goto (macro-dial-one,s,16)
    -- Executing [s@macro-dial-one:16] GotoIf("SIP/SIP_TRUNK-000076f8", "1?next2:continue") in new stack
    -- Goto (macro-dial-one,s,17)
    -- Executing [s@macro-dial-one:17] GotoIf("SIP/SIP_TRUNK-000076f8", "1?continue") in new stack
    -- Goto (macro-dial-one,s,25)
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/SIP_TRUNK-000076f8", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/SIP_TRUNK-000076f8", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/SIP_TRUNK-000076f8", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/SIP_TRUNK-000076f8", "DEVICES=1114") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Set(DEVICES=114)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/SIP_TRUNK-000076f8", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/SIP_TRUNK-000076f8", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/SIP_TRUNK-000076f8", "THISDIAL=SIP/1114") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/SIP_TRUNK-000076f8", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/SIP_TRUNK-000076f8", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/SIP_TRUNK-000076f8", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/SIP_TRUNK-000076f8", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/SIP_TRUNK-000076f8", "THISPART2=SIP/1114") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/SIP_TRUNK-000076f8", "0?Set(THISPART2=DAHDI/1114)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/SIP_TRUNK-000076f8", "NEWDIAL=SIP/1114&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/SIP_TRUNK-000076f8", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/SIP_TRUNK-000076f8", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/SIP_TRUNK-000076f8", "THISDIAL=SIP/1114") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/SIP_TRUNK-000076f8", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("SIP/SIP_TRUNK-000076f8", "DSTRING=SIP/1114&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("SIP/SIP_TRUNK-000076f8", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/SIP_TRUNK-000076f8", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("SIP/SIP_TRUNK-000076f8", "DSTRING=SIP/1114") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("SIP/SIP_TRUNK-000076f8", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/SIP_TRUNK-000076f8", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/SIP_TRUNK-000076f8", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/SIP_TRUNK-000076f8", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/SIP_TRUNK-000076f8", "DB(CALLTRACE/1114)=+4917844444188") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/SIP_TRUNK-000076f8", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("SIP/SIP_TRUNK-000076f8", "D_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/SIP_TRUNK-000076f8", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/SIP_TRUNK-000076f8", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/SIP_TRUNK-000076f8", "1?Set(CHANNEL(musicclass)=default)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/SIP_TRUNK-000076f8", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/SIP_TRUNK-000076f8", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/SIP_TRUNK-000076f8", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("SIP/SIP_TRUNK-000076f8", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("SIP/SIP_TRUNK-000076f8", "1?godial") in new stack
    -- Goto (macro-dial-one,s,42)
    -- Executing [s@macro-dial-one:42] Dial("SIP/SIP_TRUNK-000076f8", "SIP/1114,,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 18532
set_destination: Parsing <sip:1113@10.254.102.114:47693> for address/port to send to
set_destination: set destination to 10.254.102.114:47693
Reliably Transmitting (no NAT) to 10.254.102.114:47693:
NOTIFY sip:1113@10.254.102.114:47693 SIP/2.0
Via: SIP/2.0/UDP 10.254.100.170:5060;branch=z9hG4bK1b807f73;rport
Max-Forwards: 70
From: <sip:1114@10.254.100.170>;tag=as38f65599
To: <sip:1113@10.254.100.170>;tag=ip33ektz64
Contact: <sip:1114@10.254.100.170:5060>
Call-ID: 3135323735313539373337333133-rs04uvokb0xw
CSeq: 106 NOTIFY
User-Agent: FPBX-2.11.0(1.8.32.2)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 224

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4" state="full" entity="sip:1114@10.254.100.170">
<dialog id="1114" direction="recipient">
<state>early</state>
</dialog>
</dialog-info>

---
  == Extension Changed 1114[ext-local] new state Ringing for Notify User 1113
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.254.102.237:5060:
INVITE sip:1114@10.254.102.237:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.254.100.170:5060;branch=z9hG4bK44e63e04
Max-Forwards: 70
From: "+4917844444188" <sip:+4917844444188@10.254.100.170>;tag=as5dd17b36
To: <sip:1114@10.254.102.237:5060;transport=udp>
Contact: <sip:+4917844444188@10.254.100.170:5060>
Call-ID: 3346f5f64299fd2a5f82f2167a1d6e2e@10.254.100.170:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.32.2)
Date: Mon, 28 May 2018 14:41:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 126831909 126831909 IN IP4 10.254.100.170
s=Asterisk PBX 1.8.32.2
c=IN IP4 10.254.100.170
t=0 0
m=audio 18532 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/1114

<--- Transmitting (no NAT) to provider_net.136.115:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP provider_net.136.115:5060;branch=z9hG4bKa6k11810bobmaofg54c0.1;received=provider_net.136.115
From: <sip:+4917844444188@ip_sip_provider;user=phone>;tag=SDo0f9801-l9lesar1-CC-37
To: <sip:+496955551114@ip_sip_provider;user=phone>;tag=as520189d4
Call-ID: SDo0f9801-dbeec90b68019b72a92491be51913d95-l65h813
CSeq: 1 INVITE
Server: FPBX-2.11.0(1.8.32.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+496955551114@our_ip:5060;transport=TCP>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.254.102.114:47693 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.254.100.170:5060;branch=z9hG4bK1b807f73;rport=5060
From: <sip:1114@10.254.100.170>;tag=as38f65599
To: <sip:1113@10.254.100.170>;tag=ip33ektz64
Call-ID: 3135323735313539373337333133-rs04uvokb0xw
CSeq: 106 NOTIFY
User-Agent: snomD375/8.9.3.80
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.254.102.237:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.254.100.170:5060;branch=z9hG4bK44e63e04
From: "+4917844444188" <sip:+4917844444188@10.254.100.170>;tag=as5dd17b36
To: <sip:1114@10.254.102.237:5060;transport=udp>;tag=1877672511
Call-ID: 3346f5f64299fd2a5f82f2167a1d6e2e@10.254.100.170:5060
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Allow-Events: talk, hold, conference
Contact: 1114 <sip:1114@10.254.102.237:5060;transport=udp>
Server: optiPoint 410 Economy+/V7 V7 R6.10.0
X-Siemens-Call-Type: ST-insecure
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:1114@10.254.102.237:5060;transport=udp>
    -- SIP/1114-000076f9 is ringing

<--- Transmitting (no NAT) to provider_net.136.115:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP provider_net.136.115:5060;branch=z9hG4bKa6k11810bobmaofg54c0.1;received=provider_net.136.115
From: <sip:+4917844444188@ip_sip_provider;user=phone>;tag=SDo0f9801-l9lesar1-CC-37
To: <sip:+496955551114@ip_sip_provider;user=phone>;tag=as520189d4
Call-ID: SDo0f9801-dbeec90b68019b72a92491be51913d95-l65h813
CSeq: 1 INVITE
Server: FPBX-2.11.0(1.8.32.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+496955551114@our_ip:5060;transport=TCP>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '75b660ea15f5138c0e3d606273890775@10.254.100.170:5060' Method: OPTIONS
freepbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

  

this is a call coming from my cell 4917844444188 to extension 49695555114 which is in turn extension 1114. 1113 is my deskphone.


(sorry this is distributed across several posts, i did not find any option to edit my posts.)

ok, step by step.


found this:

http://wiki.snom.com/Features/Extension_Monitoring/Ringing


"The "Calls" screen is NOT triggered due to the absence of the <remote> / <local> elements."


so my guess was right. also, that means it is not something on the snom side i am looking for. anyone knows how to get asterisk put those fields into the xml-structure?


thank you,

best,

sebastian

Hello Sebastian, 


please take a look to this forum enty; It seems to deal with the same topic: 

http://forum.snom.com/index.php?showtopic=11325


Best regards

Snom Support Team

hi sidathy.


the thread over there really is about the same issue. and it is we have a custom context too. so this is the right direction. i need to see how to deal with the different contexts.


thanks so far a lot. i will post back my findings.


best,

sebastian

hi sebastian, 


you are welcome!


Best regards

snom support team

Answer

to see how things go on a default install, i started from scratch on fresh installation of freepbx 14.0.3.1 with asterisk 15.4.0.


to finalize my individual setup i changed this:

- set chan_sip to udp 5060 (i want to use chan_sip and not pjsip)

- add fiield "notifycid = yes" in chan sip settings in other sip settings.


after that i created three extensions.


on the snom phone d375 i set a function key to "blf": context to ext@freepbx_ip (and not "active"), and number to "sip:other_ext@freepbx_ip|**". the string will be constructed by the phone, i only typed into the field "other_ext|**". where freepbx_ip is something like 10.10.10.1 and other_ext is something like 1234.


and voila, everythin is running so smoothly. blf with call screen almost out of the box. very nice.


so what was different now? at first it is the contexts. in the setup above, all contexts are set to the freepbx defaults. playing around with contexts you really need to know where the calls will go and what happens behind the scenes.


second, it is the asterisk version. with the default install using freepbx 14 with asterisk 13 it did not work at all!


i hope that helps anyone with the same requirement on snom phones.


best,

sebastian


Hi Sebastian, 


many thanks for sharing the solution and your expierience with us and our customers!


Best regards

snom support team

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