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MTT.Magic Тишина при входящем вызове

0

Тишина при входящем вызове на номер MTT.Magic, ошибок вроде никаких нет. Судя по sip show channels используется кодек ulaw. Сервер находится за NATом в локальной сети, стоит Asterisk 1.8.

sip.conf

[mtt]
type=peer
context=incoming
username=74997097336
fromuser=74997097336
fromdomain=voip.mtt.ru
host=voip.mtt.ru
secret=password
nat=yes
qualify=yes
insecure=port,invite
canreinvite=no
disallow=all
allow=ulaw
allow=g722

extensions.conf

[mtt]
exten => 74997097336,1,Answer
exten => 74997097336,n,MusicOnHold()

[incoming]
include => megafon
include => mtt

asterisk -rvvv

Verbosity was 0 and is now 3
  == Using SIP RTP CoS mark 5
    -- Executing [74997097336@incoming:1] Answer("SIP/mtt-00000000", "") in new stack
    -- Executing [74997097336@incoming:2] MusicOnHold("SIP/mtt-00000000", "") in new stack
    -- Started music on hold, class 'default', on SIP/mtt-00000000
    -- Stopped music on hold on SIP/mtt-00000000
  == Spawn extension (incoming, 74997097336, 3) exited non-zero on 'SIP/mtt-00000000'
удалить закрыть спам изменить тег редактировать

спросил 2011-06-08 16:54:59 +0400

Aryutk Gravatar Aryutk
1 ● 1 ● 2

Comments

посмотри rtp set debug ip voip.mtt.ru есть трафик во время звонка alexcr ( 2011-06-08 17:01:36 +0400 )редактировать
Если да ngrep в помощь. Там будет видно кто icp-unreacible шлет aoz1 ( 2011-06-08 17:27:08 +0400 )редактировать

4 Ответа

Сорт. по » старые новые больше голосов
0

Самое удивительное что звонок иногда проходит и музыку слышно, но 50/50. Мультифон работает нормально.

ссылка удалить спам редактировать

ответил 2011-06-08 19:39:17 +0400

Aryutk Gravatar Aryutk
1 ● 1 ● 2

обновил 2011-06-08 19:42:34 +0400

0

Мрачно копать. Тем более не зная топологии сети. Вариантов мульен.

ngrep 87.238.224.113 >rtp. И посмотрите в этой каше кто и куда udp (rtp) пакеты шлет и если где блокируется в оборотку icmp пакет идет. Отам и копайте.

А для sip в ngrep -W byline рекомендую.

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ответил 2011-06-08 18:40:30 +0400

aoz1 Gravatar aoz1
43 ● 3 ● 1 ● 3
0

Все само заработало. Жду следующего косяка, чтобы debug скинуть. Быстро обрадовался, снова те же грабли...

ngrep 87.238.224.113 port 5060

debug когда работает

interface: eth0 (192.168.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
match: 87.238.224.113
######
U 87.238.224.113:5060 -> 192.168.0.106:5060
  INVITE sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238
  .224.113:5060;branch=z9hG4bK-d8754z-c5525c61f32f5030-1---d8754z-;rport..Via
  : SIP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-jwpyxi7hhluqdxyg;rport=506
  1..Max-Forwards: 69..Record-Route: <sip:87.238.224.113;lr>..Contact: "Anony
  mous"<sip:87.238.224.113:5061>..To: <sip:74997097336@87.238.224.113>..From:
   <sip:+73912532807@87.238.224.113>;tag=h5otadfh7hes5dsb.o..Call-ID: 5595083
  7086201118059@10.128.30.61..CSeq: 612 INVITE..Expires: 300..Content-Disposi
  tion: session..Content-Type: application/sdp..User-Agent: Sippy..Content-Le
  ngth: 324..cisco-GUID: 2053471671-2446725600-2449473584-1221792810..h323-co
  nf-id: 2053471671-2446725600-2449473584-1221792810....v=0..o=Sippy 15017646
  0 0 IN IP4 87.238.224.113..s=Phone-Call..t=0 0..m=audio 56288 RTP/AVP 18 8 
  0 4 101..c=IN IP4 87.238.224.111..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=n
  o..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:4 G723/8000..a=fmtp
  :4 annexa=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrec
  v..                                                                        
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d87
  54z-c5525c61f32f5030-1---d8754z-;received=87.238.224.113;rport=5060..Via: S
  IP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-jwpyxi7hhluqdxyg;rport=5061..
  Record-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.1
  13>;tag=h5otadfh7hes5dsb.o..To: <sip:74997097336@87.238.224.113>..Call-ID: 
  55950837086201118059@10.128.30.61..CSeq: 612 INVITE..Server: Asterisk PBX 1
  .8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIF
  Y, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:74997097336@192
  .168.0.106:5060>..Content-Length: 0....                                    
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  c5525c61f32f5030-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-jwpyxi7hhluqdxyg;rport=5061..Reco
  rd-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.113>;
  tag=h5otadfh7hes5dsb.o..To: <sip:74997097336@87.238.224.113>;tag=as64a2ea64
  ..Call-ID: 55950837086201118059@10.128.30.61..CSeq: 612 INVITE..Server: Ast
  erisk PBX 1.8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSC
  RIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:7499
  7097233@192.168.0.106:5060>..Content-Type: application/sdp..Content-Length:
   238....v=0..o=root 773446820 773446820 IN IP4 192.168.0.106..s=Asterisk PB
  X 1.8.4.2-1..c=IN IP4 192.168.0.106..t=0 0..m=audio 19648 RTP/AVP 0 101..a=
  rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=p
  time:20..a=sendrecv..                                                      
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  c5525c61f32f5030-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-jwpyxi7hhluqdxyg;rport=5061..Reco
  rd-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.113>;
  tag=h5otadfh7hes5dsb.o..To: <sip:74997097336@87.238.224.113>;tag=as64a2ea64
  ..Call-ID: 55950837086201118059@10.128.30.61..CSeq: 612 INVITE..Server: Ast
  erisk PBX 1.8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSC
  RIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:7499
  7097233@192.168.0.106:5060>..Content-Type: application/sdp..Content-Length:
   238....v=0..o=root 773446820 773446820 IN IP4 192.168.0.106..s=Asterisk PB
  X 1.8.4.2-1..c=IN IP4 192.168.0.106..t=0 0..m=audio 19648 RTP/AVP 0 101..a=
  rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=p
  time:20..a=sendrecv..                                                      
#
U 87.238.224.113:5060 -> 192.168.0.106:5060
  ACK sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-9e62fa231e5d6839-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;rport=5061;branch=z9hG4bK-v3tvhrq755g6f76f..
  Max-Forwards: 69..To: <sip:74997097336@87.238.224.113>;tag=as64a2ea64..From
  : <sip:+73912532807@87.238.224.113>;tag=h5otadfh7hes5dsb.o..Call-ID: 559508
  37086201118059@10.128.30.61..CSeq: 612 ACK..User-Agent: Sippy..Content-Leng
  th: 0....                                                                  
#
U 87.238.224.113:5060 -> 192.168.0.106:5060
  ACK sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-ea33e21bcbb52d00-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;rport=5061;branch=z9hG4bK-v3tvhrq755g6f76f..
  Max-Forwards: 69..To: <sip:74997097336@87.238.224.113>;tag=as64a2ea64..From
  : <sip:+73912532807@87.238.224.113>;tag=h5otadfh7hes5dsb.o..Call-ID: 559508
  37086201118059@10.128.30.61..CSeq: 612 ACK..User-Agent: Sippy..Content-Leng
  th: 0....                                                                  
#
U 87.238.224.113:5060 -> 192.168.0.106:5060
  BYE sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-5b741d2a3b05fe40-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-nd3pap3qlnl7nsaj;rport=5061..
  Max-Forwards: 69..Contact: "Anonymous"<sip:87.238.224.113:5061>..To: <sip:7
  4997097233@87.238.224.113>;tag=as64a2ea64..From: <sip:+73912532807@87.238.2
  24.113>;tag=h5otadfh7hes5dsb.o..Call-ID: 55950837086201118059@10.128.30.61.
  .CSeq: 613 BYE..User-Agent: Sippy..Content-Length: 0..cisco-GUID: 205347167
  1-2446725600-2449473584-1221792810..h323-conf-id: 2053471671-2446725600-244
  9473584-1221792810....                                                     
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  5b741d2a3b05fe40-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-nd3pap3qlnl7nsaj;rport=5061..From
  : <sip:+73912532807@87.238.224.113>;tag=h5otadfh7hes5dsb.o..To: <sip:749970
  97233@87.238.224.113>;tag=as64a2ea64..Call-ID: 55950837086201118059@10.128.
  30.61..CSeq: 613 BYE..Server: Asterisk PBX 1.8.4.2-1..Allow: INVITE, ACK, C
  ANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re
  places, timer..Content-Length: 0....                                       
####^Cexit
18 received, 0 dropped

debug не работает

interface: eth0 (192.168.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
match: 87.238.224.113
###
U 87.238.224.113:5060 -> 192.168.0.106:5060
  INVITE sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238
  .224.113:5060;branch=z9hG4bK-d8754z-87af3e6db39df65b-1---d8754z-;rport..Via
  : SIP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-5povzp7s6fnhm7b3;rport=506
  1..Max-Forwards: 69..Record-Route: <sip:87.238.224.113;lr>..Contact: "Anony
  mous"<sip:87.238.224.113:5061>..To: <sip:74997097336@87.238.224.113>..From:
   <sip:+73912532807@87.238.224.113>;tag=2rzruyfix6v7lvqh.o..Call-ID: 7150478
  53862011181159@10.128.30.61..CSeq: 870 INVITE..Expires: 300..Content-Dispos
  ition: session..Content-Type: application/sdp..User-Agent: Sippy..Content-L
  ength: 324..cisco-GUID: 63799956-2446856672-2449473584-1221792810..h323-con
  f-id: 63799956-2446856672-2449473584-1221792810....v=0..o=Sippy 143398220 0
   IN IP4 87.238.224.113..s=Phone-Call..t=0 0..m=audio 58914 RTP/AVP 18 8 0 4
   101..c=IN IP4 87.238.224.111..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..
  a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:4 G723/8000..a=fmtp:4 
  annexa=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d87
  54z-87af3e6db39df65b-1---d8754z-;received=87.238.224.113;rport=5060..Via: S
  IP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-5povzp7s6fnhm7b3;rport=5061..
  Record-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.1
  13>;tag=2rzruyfix6v7lvqh.o..To: <sip:74997097336@87.238.224.113>..Call-ID: 
  715047853862011181159@10.128.30.61..CSeq: 870 INVITE..Server: Asterisk PBX 
  1.8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI
  FY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:74997097336@19
  2.168.0.106:5060>..Content-Length: 0....                                   
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  87af3e6db39df65b-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-5povzp7s6fnhm7b3;rport=5061..Reco
  rd-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.113>;
  tag=2rzruyfix6v7lvqh.o..To: <sip:74997097336@87.238.224.113>;tag=as13303b17
  ..Call-ID: 715047853862011181159@10.128.30.61..CSeq: 870 INVITE..Server: As
  terisk PBX 1.8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBS
  CRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:749
  97097233@192.168.0.106:5060>..Content-Type: application/sdp..Content-Length
  : 240....v=0..o=root 1060981597 1060981597 IN IP4 192.168.0.106..s=Asterisk
   PBX 1.8.4.2-1..c=IN IP4 192.168.0.106..t=0 0..m=audio 16428 RTP/AVP 0 101.
  .a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..
  a=ptime:20..a=sendrecv..                                                   
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  87af3e6db39df65b-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-5povzp7s6fnhm7b3;rport=5061..Reco
  rd-Route: <sip:87.238.224.113;lr>..From: <sip:+73912532807@87.238.224.113>;
  tag=2rzruyfix6v7lvqh.o..To: <sip:74997097336@87.238.224.113>;tag=as13303b17
  ..Call-ID: 715047853862011181159@10.128.30.61..CSeq: 870 INVITE..Server: As
  terisk PBX 1.8.4.2-1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBS
  CRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:749
  97097233@192.168.0.106:5060>..Content-Type: application/sdp..Content-Length
  : 240....v=0..o=root 1060981597 1060981597 IN IP4 192.168.0.106..s=Asterisk
   PBX 1.8.4.2-1..c=IN IP4 192.168.0.106..t=0 0..m=audio 16428 RTP/AVP 0 101.
  .a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..
  a=ptime:20..a=sendrecv..                                                   
#
U 87.238.224.113:5060 -> 192.168.0.106:5060
  ACK sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-23da903d498c7d64-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;rport=5061;branch=z9hG4bK-5j5hb3vi3ob5bnim..
  Max-Forwards: 69..To: <sip:74997097336@87.238.224.113>;tag=as13303b17..From
  : <sip:+73912532807@87.238.224.113>;tag=2rzruyfix6v7lvqh.o..Call-ID: 715047
  853862011181159@10.128.30.61..CSeq: 870 ACK..User-Agent: Sippy..Content-Len
  gth: 0....                                                                 
#
U 87.238.224.113:5060 -> 192.168.0.106:5060
  ACK sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-b91d5d6031c3371f-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;rport=5061;branch=z9hG4bK-5j5hb3vi3ob5bnim..
  Max-Forwards: 69..To: <sip:74997097336@87.238.224.113>;tag=as13303b17..From
  : <sip:+73912532807@87.238.224.113>;tag=2rzruyfix6v7lvqh.o..Call-ID: 715047
  853862011181159@10.128.30.61..CSeq: 870 ACK..User-Agent: Sippy..Content-Len
  gth: 0....                                                                 
####
U 87.238.224.113:5060 -> 192.168.0.106:5060
  BYE sip:74997097336@89.105.150.137:5060 SIP/2.0..Via: SIP/2.0/UDP 87.238.22
  4.113:5060;branch=z9hG4bK-d8754z-4cb0160d655f7e17-1---d8754z-;rport..Via: S
  IP/2.0/UDP 87.238.224.113:5061;branch=z9hG4bK-7unrzk7bmvkzfiz7;rport=5061..
  Max-Forwards: 69..Contact: "Anonymous"<sip:87.238.224.113:5061>..To: <sip:7
  4997097233@87.238.224.113>;tag=as13303b17..From: <sip:+73912532807@87.238.2
  24.113>;tag=2rzruyfix6v7lvqh.o..Call-ID: 715047853862011181159@10.128.30.61
  ..CSeq: 871 BYE..User-Agent: Sippy..Content-Length: 0..cisco-GUID: 63799956
  -2446856672-2449473584-1221792810..h323-conf-id: 63799956-2446856672-244947
  3584-1221792810....                                                        
#
U 192.168.0.106:5060 -> 87.238.224.113:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 87.238.224.113:5060;branch=z9hG4bK-d8754z-
  4cb0160d655f7e17-1---d8754z-;received=87.238.224.113;rport=5060..Via: SIP/2
  .0/UDP 87.238.224.113:5061;branch=z9hG4bK-7unrzk7bmvkzfiz7;rport=5061..From
  : <sip:+73912532807@87.238.224.113>;tag=2rzruyfix6v7lvqh.o..To: <sip:749970
  97233@87.238.224.113>;tag=as13303b17..Call-ID: 715047853862011181159@10.128
  .30.61..CSeq: 871 BYE..Server: Asterisk PBX 1.8.4.2-1..Allow: INVITE, ACK, 
  CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: r
  eplaces, timer..Content-Length: 0....                                      
###^Cexit
17 received, 0 dropped
ссылка удалить спам редактировать

ответил 2011-06-08 17:54:36 +0400

Aryutk Gravatar Aryutk
1 ● 1 ● 2

обновил 2011-06-08 18:03:29 +0400

0

Мм, интересно. RTP приходит с IP-адреса 87.238.224.111, а не с 87.238.224.113.

ссылка удалить спам редактировать

ответил 2011-06-08 19:11:57 +0400

Aryutk Gravatar Aryutk
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Задан: 2011-06-08 16:54:59 +0400

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Обновлен: Jun 08 '11

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.