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Multifon + Asterisk(FreePBX) + DTMF

https://forum.asterisk.ru/viewtopic.php?t=7087

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Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 10:58
Dangerous13
Поставили задачу мигрировать телефонию с астериска 1.4.40.1 на 11.15 или выше. Был установлен дистриб Asterisk+Elastix+FreePBX. Перенесли все транки/номера и т.п. Потихоньку начали переводить людей на новый сервер. И при переводе очередной группы возникла проблема. Проблема с ТРАНКом от Мегафона. А именно с донабором номера.
Задача: Звонок на мегафонский номер направляется на DISA, дальше набирается номер очереди и звонок поступает кому надо. Но на новом сервере при попытке донабора ничего не дозванивается, DISA обрывает связь по таймауту.
Поменял направление вызова с DISA на IVR, то-же самое, не понимает цифры, по не ответу попадаем на нужную очередь. Но связь односторонняя, слышно звонящего но не слышно принявшего звонок.
Направление на внутренний номер работает отлично, связь двухсторонняя но задержка голоса примерно в секунду, полторы.

Все другие ТРАНКи с телефонной станции принимаются отлично, DISA и IVR обрабатываются без проблем.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: PEER

Код: Выделить всё

    username=<7ТЕЛЕФОН>
    type=peer
    secret=<ПАРОЛЬ>
    qualify=yes
    nat=yes
    insecure=very
    host=sbc.megafon.ru
    fromuser=<7ТЕЛЕФОН>
    dtmfmode=inband
    disallow=all
    context=from-trunk
    canreinvite=no
    authname=<7ТЕЛЕФОН>
    allow=ulaw
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: USER

Код: Выделить всё

    type=user
    secret=<ПАРОЛЬ>
    context=from-trunk
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: LOG

Код: Выделить всё

      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Executing [792756XXXXX@from-trunk:1] Set("SIP/9XXXXX_in-00000092", "__FROM_DID=792756XXXXX") in new stack
        -- Executing [792756XXXXX@from-trunk:2] Set("SIP/9XXXXX_in-00000092", "__REC_POLICY_MODE=always") in new stack
        -- Executing [792756XXXXX@from-trunk:3] Gosub("SIP/9XXXXX_in-00000092", "app-blacklist-check,s,1()") in new stack
        -- Executing [s@app-blacklist-check:1] GotoIf("SIP/9XXXXX_in-00000092", "0?blacklisted") in new stack
        -- Executing [s@app-blacklist-check:2] Set("SIP/9XXXXX_in-00000092", "CALLED_BLACKLIST=1") in new stack
        -- Executing [s@app-blacklist-check:3] Return("SIP/9XXXXX_in-00000092", "") in new stack
        -- Executing [792756XXXXX@from-trunk:4] Gosub("SIP/9XXXXX_in-00000092", "cidlookup,cidlookup_1,1()") in new stack
        -- Executing [cidlookup_1@cidlookup:1] Set("SIP/9XXXXX_in-00000092", "CURLOPT(httptimeout)=7") in new stack
        -- Executing [cidlookup_1@cidlookup:2] Set("SIP/9XXXXX_in-00000092", "CALLERID(name)=A134113+ A137787+B") in new stack
        -- Executing [cidlookup_1@cidlookup:3] Return("SIP/9XXXXX_in-00000092", "") in new stack
        -- Executing [792756XXXXX@from-trunk:5] Set("SIP/9XXXXX_in-00000092", "CDR(did)=792756XXXXX") in new stack
        -- Executing [792756XXXXX@from-trunk:6] ExecIf("SIP/9XXXXX_in-00000092", "0 ?Set(CALLERID(name)=78512XXXXXX)") in new stack
        -- Executing [792756XXXXX@from-trunk:7] Set("SIP/9XXXXX_in-00000092", "CHANNEL(musicclass)=NewYear") in new stack
        -- Executing [792756XXXXX@from-trunk:8] Set("SIP/9XXXXX_in-00000092", "__MOHCLASS=NewYear") in new stack
        -- Executing [792756XXXXX@from-trunk:9] Set("SIP/9XXXXX_in-00000092", "__CALLINGPRES_SV=allowed_not_screened") in new stack
        -- Executing [792756XXXXX@from-trunk:10] Set("SIP/9XXXXX_in-00000092", "CALLERPRES()=allowed_not_screened") in new stack
        -- Executing [792756XXXXX@from-trunk:11] Goto("SIP/9XXXXX_in-00000092", "disa,1,1") in new stack
        -- Goto (disa,1,1)
        -- Executing [1@disa:1] Answer("SIP/9XXXXX_in-00000092", "") in new stack
      == Spawn extension (disa, 1, 1) exited non-zero on 'SIP/9XXXXX_in-00000092'

В чём может быть проблема. dtmfmode пробовал разные, результат один, не работает.

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 11:49
Dangerous13
Дополнительно Debug:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Debig

Код: Выделить всё

<--- SIP read from UDP:193.201.229.35:5060 --->
INVITE sip:792756XXXXX@10.1.0.95:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK87rdnd00d02jvr63d100.1
Max-Forwards: 68
User-Agent: multifon.ru
Supported: 100rel,path,replaces,tdialog
Expires: 300
Content-Type: application/sdp
Content-Length: 293
From: <sip:798859XXXXX@multifon.ru>;tag=SD41pm901-69853246313536414DF53E07
To: sip:792756XXXXX@multifon.ru:5060
P-Asserted-Identity: sip:798859XXXXX@multifon.ru:5060
Allow: PRACK,INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER
Call-ID: SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00
CSeq: 1 INVITE
Contact: <sip:798859XXXXX@193.201.229.35:5060;transport=udp>

v=0
o=Dialogic_SDP 3237687 0 IN IP4 193.201.229.19
s=Dialogic-SIP
c=IN IP4 193.201.229.19
t=0 0
m=audio 14222 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=silenceSupp:off - - - -
<------------->
--- (15 headers 13 lines) ---
Sending to 193.201.229.35:5060 (no NAT)
Sending to 193.201.229.35:5060 (no NAT)
Using INVITE request as basis request - SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00
Found peer '9XXXXX_in' for '798859XXXXX' from 193.201.229.35:5060
  == 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 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
Capabilities: us - (ulaw|alaw|g726|g729), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.201.229.19:14222
Looking for 792756XXXXX in from-trunk (domain 10.1.0.95)
list_route: hop: <sip:798859XXXXX@193.201.229.35:5060;transport=udp>

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK87rdnd00d02jvr63d100.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD41pm901-69853246313536414DF53E07
To: sip:792756XXXXX@multifon.ru:5060
Call-ID: SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Length: 0


<------------>
    -- Executing [792756XXXXX@from-trunk:1] Set("SIP/9XXXXX_in-000003c2", "__FROM_DID=792756XXXXX") in new stack
    -- Executing [792756XXXXX@from-trunk:2] Set("SIP/9XXXXX_in-000003c2", "__REC_POLICY_MODE=always") in new stack
    -- Executing [792756XXXXX@from-trunk:3] Gosub("SIP/9XXXXX_in-000003c2", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/9XXXXX_in-000003c2", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/9XXXXX_in-000003c2", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/9XXXXX_in-000003c2", "") in new stack
    -- Executing [792756XXXXX@from-trunk:4] Gosub("SIP/9XXXXX_in-000003c2", "cidlookup,cidlookup_1,1()") in new stack
    -- Executing [cidlookup_1@cidlookup:1] Set("SIP/9XXXXX_in-000003c2", "CURLOPT(httptimeout)=7") in new stack
    -- Executing [cidlookup_1@cidlookup:2] Set("SIP/9XXXXX_in-000003c2", "CALLERID(name)=A6370-B A187346-B") in new stack
    -- Executing [cidlookup_1@cidlookup:3] Return("SIP/9XXXXX_in-000003c2", "") in new stack
    -- Executing [792756XXXXX@from-trunk:5] Set("SIP/9XXXXX_in-000003c2", "CDR(did)=792756XXXXX") in new stack
    -- Executing [792756XXXXX@from-trunk:6] ExecIf("SIP/9XXXXX_in-000003c2", "0 ?Set(CALLERID(name)=798859XXXXX)") in new stack
    -- Executing [792756XXXXX@from-trunk:7] Set("SIP/9XXXXX_in-000003c2", "CHANNEL(musicclass)=NewYear") in new stack
    -- Executing [792756XXXXX@from-trunk:8] Set("SIP/9XXXXX_in-000003c2", "__MOHCLASS=NewYear") in new stack
    -- Executing [792756XXXXX@from-trunk:9] Ringing("SIP/9XXXXX_in-000003c2", "") in new stack

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK87rdnd00d02jvr63d100.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD41pm901-69853246313536414DF53E07
To: sip:792756XXXXX@multifon.ru:5060;tag=as4275cfcc
Call-ID: SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Length: 0


<------------>
    -- Executing [792756XXXXX@from-trunk:10] Set("SIP/9XXXXX_in-000003c2", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [792756XXXXX@from-trunk:11] Set("SIP/9XXXXX_in-000003c2", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [792756XXXXX@from-trunk:12] Goto("SIP/9XXXXX_in-000003c2", "disa,1,1") in new stack
    -- Goto (disa,1,1)
    -- Executing [1@disa:1] Answer("SIP/9XXXXX_in-000003c2", "") in new stack
Audio is at 10020
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP

<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK87rdnd00d02jvr63d100.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD41pm901-69853246313536414DF53E07
To: sip:792756XXXXX@multifon.ru:5060;tag=as4275cfcc
Call-ID: SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 832789589 832789589 IN IP4 10.1.0.95
s=Asterisk PBX 11.15.0
c=IN IP4 10.1.0.95
t=0 0
m=audio 10020 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:193.201.229.35:5060 --->
ACK sip:792756XXXXX@10.1.0.95:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK69l6a4102gvl3l3r3i70.1
Max-Forwards: 68
User-Agent: multifon.ru
Content-Length: 0
CSeq: 1 ACK
To: sip:792756XXXXX@multifon.ru:5060;tag=as4275cfcc
From: <sip:798859XXXXX@multifon.ru>;tag=SD41pm901-69853246313536414DF53E07
Call-ID: SD41pm901-149f3b3069ccc2cbe8044c412600f27e-v300g00

<------------->
--- (9 headers 0 lines) ---
    -- Executing [1@disa:2] Set("SIP/9XXXXX_in-000003c2", "_DISA=disa^1^newcall") in new stack
    -- Executing [1@disa:3] Set("SIP/9XXXXX_in-000003c2", "_DISACONTEXT=from-internal") in new stack
    -- Executing [1@disa:4] Set("SIP/9XXXXX_in-000003c2", "_KEEPCID=TRUE") in new stack
    -- Executing [1@disa:5] Set("SIP/9XXXXX_in-000003c2", "_HANGUP=") in new stack
    -- Executing [1@disa:6] Set("SIP/9XXXXX_in-000003c2", "TIMEOUT(digit)=2") in new stack
    -- Digit timeout set to 2.000
    -- Executing [1@disa:7] Set("SIP/9XXXXX_in-000003c2", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10.000
    -- Executing [1@disa:8] DISA("SIP/9XXXXX_in-000003c2", "no-password,disa-dial") in new stack

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 12:01
awsswa
не надо создавать темы на всех форумах где смогли зарегистрироваться - везде одни и теже люди
Проще побегать по гуглу лишнии пару часов
Надеюсь вы не думаете что до вас никто подобных проблем не решал ?

http://asterisk-support.ru/question/607 ... epbx-dtmf/

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 12:25
ded
Dialogic от Мультифона вообще не предлагает никаких вариантов передачи DTMF (обычно стандартно идёт outband 101 = rfc 2833), только 8 0 18 4 - ulaw, alaw, g729(annex b) и g723 (с выключенной опцией подавления тишины):

Код: Выделить всё

v=0
o=Dialogic_SDP 3237687 0 IN IP4 193.201.229.19
s=Dialogic-SIP
c=IN IP4 193.201.229.19
t=0 0
m=audio 14222 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=silenceSupp:off
на вашей стороне пробуйте ограничить только ulaw, alaw и пытаться принимать в inband. Включите дебаг DTMF

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 12:54
awsswa
он по ходу рестарт не делает
потому как в логах все кодаки подтверждает - а в настройках только ulaw

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 13:28
Dangerous13
И рестарты делал и сервак перезагружал. Не помогает. Если не использовать DISA или IVR то всё замечательно. Но нужно чтобы звонок приходил на DISA.

Текущие параметры:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: peer

Код: Выделить всё

username=792756XXXXX
type=peer
secret=XXXXX
qualify=yes
nat=yes
[b]insecure=port,invite[/b]
host=sbc.megafon.ru
fromuser=792756XXXXX
[b]dtmfmode=inband[/b]
disallow=all
context=from-trunk
canreinvite=no
authname=792756XXXXX
[b]allow=ulaw[/b]
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: DEBUG

Код: Выделить всё

<--- SIP read from UDP:193.201.229.35:5060 --->
INVITE sip:792756XXXXX@10.1.0.95:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK0q21k510206ob9319b60.1
Max-Forwards: 68
User-Agent: multifon.ru
Supported: 100rel,path,replaces,tdialog
Expires: 300
Content-Type: application/sdp
Content-Length: 293
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
To: sip:792756XXXXX@multifon.ru:5060
P-Asserted-Identity: sip:798859XXXXX@multifon.ru:5060
Allow: PRACK,INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
CSeq: 1 INVITE
Contact: <sip:798859XXXXX@193.201.229.35:5060;transport=udp>

v=0
o=Dialogic_SDP 3248009 0 IN IP4 193.201.229.19
s=Dialogic-SIP
c=IN IP4 193.201.229.19
t=0 0
m=audio 10676 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=silenceSupp:off - - - -
<------------->
--- (15 headers 13 lines) ---
Sending to 193.201.229.35:5060 (no NAT)
Sending to 193.201.229.35:5060 (no NAT)
Using INVITE request as basis request - SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
Found peer '9XXXXX_in' for '798859XXXXX' from 193.201.229.35:5060
  == 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 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
Capabilities: us - (ulaw), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.201.229.19:10676
Looking for 792756XXXXX in from-trunk (domain 10.1.0.95)
list_route: hop: <sip:798859XXXXX@193.201.229.35:5060;transport=udp>

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK0q21k510206ob9319b60.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
To: sip:792756XXXXX@multifon.ru:5060
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Length: 0


<------------>
    -- Executing [792756XXXXX@from-trunk:1] Set("SIP/9XXXXX_in-00000020", "__FROM_DID=792756XXXXX") in new stack
    -- Executing [792756XXXXX@from-trunk:2] Set("SIP/9XXXXX_in-00000020", "__REC_POLICY_MODE=always") in new stack
    -- Executing [792756XXXXX@from-trunk:3] Gosub("SIP/9XXXXX_in-00000020", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/9XXXXX_in-00000020", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/9XXXXX_in-00000020", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/9XXXXX_in-00000020", "") in new stack
    -- Executing [792756XXXXX@from-trunk:4] Gosub("SIP/9XXXXX_in-00000020", "cidlookup,cidlookup_1,1()") in new stack
    -- Executing [cidlookup_1@cidlookup:1] Set("SIP/9XXXXX_in-00000020", "CURLOPT(httptimeout)=7") in new stack
    -- Executing [cidlookup_1@cidlookup:2] Set("SIP/9XXXXX_in-00000020", "CALLERID(name)=A6370-B A187346-B") in new stack
    -- Executing [cidlookup_1@cidlookup:3] Return("SIP/9XXXXX_in-00000020", "") in new stack
    -- Executing [792756XXXXX@from-trunk:5] Set("SIP/9XXXXX_in-00000020", "CDR(did)=792756XXXXX") in new stack
    -- Executing [792756XXXXX@from-trunk:6] ExecIf("SIP/9XXXXX_in-00000020", "0 ?Set(CALLERID(name)=798859XXXXX)") in new stack
    -- Executing [792756XXXXX@from-trunk:7] Set("SIP/9XXXXX_in-00000020", "CHANNEL(musicclass)=NewYear") in new stack
    -- Executing [792756XXXXX@from-trunk:8] Set("SIP/9XXXXX_in-00000020", "__MOHCLASS=NewYear") in new stack
    -- Executing [792756XXXXX@from-trunk:9] Ringing("SIP/9XXXXX_in-00000020", "") in new stack

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK0q21k510206ob9319b60.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
To: sip:792756XXXXX@multifon.ru:5060;tag=as250a6fd4
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Length: 0


<------------>
    -- Executing [792756XXXXX@from-trunk:10] Set("SIP/9XXXXX_in-00000020", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [792756XXXXX@from-trunk:11] Set("SIP/9XXXXX_in-00000020", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [792756XXXXX@from-trunk:12] Goto("SIP/9XXXXX_in-00000020", "disa,1,1") in new stack
    -- Goto (disa,1,1)
    -- Executing [1@disa:1] Answer("SIP/9XXXXX_in-00000020", "") in new stack
Audio is at 11404
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK0q21k510206ob9319b60.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
To: sip:792756XXXXX@multifon.ru:5060;tag=as250a6fd4
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:792756XXXXX@10.1.0.95:5060>
Content-Type: application/sdp
Content-Length: 172

v=0
o=root 406989993 406989993 IN IP4 10.1.0.95
s=Asterisk PBX 11.15.0
c=IN IP4 10.1.0.95
t=0 0
m=audio 11404 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:193.201.229.35:5060 --->
ACK sip:792756XXXXX@10.1.0.95:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bKk3rrnk105om4cju2tnr0.1
CSeq: 1 ACK
To: sip:792756XXXXX@multifon.ru:5060;tag=as250a6fd4
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
Max-Forwards: 69
Content-Length: 0

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

<--- SIP read from UDP:193.201.229.35:5060 --->
BYE sip:792756XXXXX@10.1.0.95:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bKk3rrnk105om4cju2tnr0cd0000010.1
To: sip:792756XXXXX@multifon.ru:5060;tag=as250a6fd4
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
Max-Forwards: 69
CSeq: 2 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 193.201.229.35:5060 (NAT)
Scheduling destruction of SIP dialog 'SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bKk3rrnk105om4cju2tnr0cd0000010.1;received=193.201.229.35;rport=5060
From: <sip:798859XXXXX@multifon.ru>;tag=SD2plo701-74F6324631353641A4B64807
To: sip:792756XXXXX@multifon.ru:5060;tag=as250a6fd4
Call-ID: SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020
CSeq: 2 BYE
Server: FPBX-2.11.0(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (disa, 1, 1) exited non-zero on 'SIP/9XXXXX_in-00000020'
Really destroying SIP dialog 'SD2plo701-1f4008919be396a2002bbec6742e9a65-v300g00020' Method: BYE
Смущает следующая запись: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 13:39
awsswa
Я что то в логах совсем не увидал попыток нажатий клавиш - у вас логи по DTMF включены ?

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 13:49
Dangerous13
Вроде как включены. По крайней мере если я совершаю звонок с софт-фона подключенного к этой телефонии в логах:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: log

Код: Выделить всё

[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4216 __ast_read: DTMF begin '1' received on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4216 __ast_read: DTMF begin '1' received on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4227 __ast_read: DTMF begin passthrough '1' on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4227 __ast_read: DTMF begin passthrough '1' on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4130 __ast_read: DTMF end '1' received on SIP/953-0000013f, duration 100 ms
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4130 __ast_read: DTMF end '1' received on SIP/953-0000013f, duration 100 ms
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4171 __ast_read: DTMF end accepted with begin '1' on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4171 __ast_read: DTMF end accepted with begin '1' on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4186 __ast_read: DTMF end '1' detected to have actual duration 79 on the wire, emulation will be triggered on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4186 __ast_read: DTMF end '1' detected to have actual duration 79 on the wire, emulation will be triggered on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4193 __ast_read: DTMF end '1' has duration 79 but want minimum 80, emulating on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4193 __ast_read: DTMF end '1' has duration 79 but want minimum 80, emulating on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4250 __ast_read: DTMF end emulation of '1' queued on SIP/953-0000013f
[2015-12-15 09:47:53] DTMF[13095][C-0000004f]: channel.c:4250 __ast_read: DTMF end emulation of '1' queued on SIP/953-0000013f
Если позвонить на другой номер с сотового и в IVR выбрать пункт то лог высвечивается и отрабатывается:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: DTMF_IVR_LOG

Код: Выделить всё

adtvelasix*CLI> 
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4216 __ast_read: DTMF begin '1' received on SIP/480058_in-00000083
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4216 __ast_read: DTMF begin '1' received on SIP/480058_in-00000083
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4220 __ast_read: DTMF begin ignored '1' on SIP/480058_in-00000083
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4220 __ast_read: DTMF begin ignored '1' on SIP/480058_in-00000083
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4130 __ast_read: DTMF end '1' received on SIP/480058_in-00000083, duration 280 ms
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4130 __ast_read: DTMF end '1' received on SIP/480058_in-00000083, duration 280 ms
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4200 __ast_read: DTMF end passthrough '1' on SIP/480058_in-00000083
[2015-12-15 13:47:55] DTMF[25782][C-0000002d]: channel.c:4200 __ast_read: DTMF end passthrough '1' on SIP/480058_in-00000083

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 13:57
ded
Повторяюсь:
кто присылает инвайт?

Код: Выделить всё

v=0
o=Dialogic_SDP 3248009 0 IN IP4 193.201.229.19
s=Dialogic-SIP
Что в инвайте на тему codec capability?
m=audio 10676 RTP/AVP 8 0 18 4
и расшифровка цифр 8 0 18 4:

Код: Выделить всё

a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=silenceSupp:off
Поэтому
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

Re: Multifon + Asterisk(FreePBX) + DTMF

Добавлено: 15 дек 2015, 14:16
Dangerous13
По видимому инвайт присылает мегафон.
Может быть моя проблема связана с тем, что сервер за NAT`om?
Часовой пояс: UTC+03:00
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