asterisk.ru

Русский форум Asterisk сообщества
https://forum.asterisk.ru/

Нет звука при входящем звонке

https://forum.asterisk.ru/viewtopic.php?t=4499

Страница 1 из 3

Нет звука при входящем звонке

Добавлено: 26 июн 2014, 14:32
Yazov
Приветствую всех присутствующих! уважаемые форумчане!
После переезда на новую машину столкнулся я с проблемой - на одном из четырех направлений абонент не слышит оператора при входящем звонке. То есть мне приходит звонок, я беру трубку и меня не слышат, а я при этом слышу. Хотя при исходящем звонке - все прекрасно. Причем такое только с один провайдером. Думал проблема в кодеках - менял их и так и сяк. Софтфон Zoiper (платная которая), там пробовал выставить только G729, или только alaw. Аналогично с транком. Ничего не меняется.
И еще одна заметка - происходит это абсолютно рандомно. Кое как взял дебаг с подобного звонка, потому что 1 к 10 что такое происходит. Иногда все отлично.
Вот тот самый дебаг:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Debug

Код: Выделить всё

<--- SIP read from UDP:195.47.255.119:5060 --->
INVITE sip:+77787460016@195.189.29.18:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK5trpdi0010mhqv8gn421.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>
Max-Forwards: 69
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49489 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 432

v=0
o=- 5490278 5490278 IN IP4 195.47.255.119
s=-
c=IN IP4 195.47.255.119
t=0 0
a=sendrecv
m=audio 46160 RTP/AVP 8 96 18 97
c=IN IP4 195.47.255.119
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (13 headers 18 lines) ---
Sending to 195.47.255.119:5060 (NAT)
Sending to 195.47.255.119:5060 (NAT)
Using INVITE request as basis request - SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
Found peer 'Kcell2' for '7078121170' from 195.47.255.119:5060
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 18
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found unknown media description format AMR for ID 96
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 195.47.255.119:46160
Looking for +77787460016 in from-trunk (domain 195.189.29.18)
list_route: hop: <sip:172.16.240.20:5060;transport=udp>

<--- Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK5trpdi0010mhqv8gn421.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49489 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Length: 0


<------------>
Audio is at 10716
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK5trpdi0010mhqv8gn421.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49489 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Type: application/sdp
Content-Length: 275

v=0
o=Cisco-SIPUA 338296292 338296292 IN IP4 195.189.29.18
s=SIP Call
c=IN IP4 195.189.29.18
t=0 0
m=audio 10716 RTP/AVP 8 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:195.47.255.119:5060 --->
ACK sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK9535a8203850hr47f4n0.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Max-Forwards: 69
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49489 ACK
Content-Length: 0

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

<--- SIP read from UDP:195.47.255.119:5060 --->
INVITE sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK9535a8203850hr47f4n0cb0002c91.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Max-Forwards: 69
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49490 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 237

v=0
o=- 5490278 5490279 IN IP4 195.47.255.119
s=-
c=IN IP4 195.47.255.119
t=0 0
a=sendrecv
m=audio 46160 RTP/AVP 8 97
c=IN IP4 195.47.255.119
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (12 headers 12 lines) ---
Sending to 195.47.255.119:5060 (NAT)
Found RTP audio format 8
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 195.47.255.119:46160

<--- Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK9535a8203850hr47f4n0cb0002c91.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49490 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Length: 0


<------------>
Audio is at 10716
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK9535a8203850hr47f4n0cb0002c91.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49490 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Type: application/sdp
Content-Length: 228

v=0
o=Cisco-SIPUA 338296292 338296293 IN IP4 195.189.29.18
s=SIP Call
c=IN IP4 195.189.29.18
t=0 0
m=audio 10716 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:195.47.255.119:5060 --->
ACK sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bKa5g9eq203oo03u4j50c0.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD57kv201-4167037069
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as65693e5b
Max-Forwards: 69
Call-ID: SD57kv201-0666028ed7ee2a636171d1a69c97ed8e-v300g00
CSeq: 49490 ACK
Content-Length: 0

<------------->
У меня есть подозрения на мой IPtables:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: iptables

Код: Выделить всё

echo "1" > /proc/sys/net/ipv4/ip_forward
iptables -F
iptables -F -t filter
iptables -F -t mangle
iptables -F -t nat
# Установка политик по умолчанию
iptables -P INPUT DROP
iptables -P FORWARD DROP
iptables -P OUTPUT ACCEPT
iptables -A INPUT -i lo -j ACCEPT
iptables -A INPUT -i lo -j ACCEPT
iptables -A INPUT -i eth0 -j ACCEPT

iptables -A INPUT -i eth1 -p tcp -m tcp --dport 3875 --tcp-flags FIN,SYN,RST,ACK SYN -m recent --set --name root --rsource
iptables -A INPUT -i eth1 -p tcp -m tcp --dport 3875 -j ACCEPT

#ПРОВАЙДЕРЫ
iptables -A INPUT -s 212.154.128.94 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 212.154.128.94 -p udp -m udp --dport 10000:20000 -j ACCEPT
iptables -A INPUT -s 217.15.180.50 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 217.15.180.50 -p udp -m udp --dport 10000:20000 -j ACCEPT
iptables -A INPUT -s 195.47.255.212 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 195.47.255.212 -p udp -m udp --dport 10000:20000 -j ACCEPT
iptables -A INPUT -s 195.226.208.11 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 195.226.208.11 -p udp -m udp --dport 10000:20000 -j ACCEPT
iptables -A INPUT -s 195.47.255.119 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 195.47.255.119 -p udp -m udp --dport 10000:20000 -j ACCEPT
iptables -A INPUT -s 172.16.240.20 -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -s 172.16.240.20 -p udp -m udp --dport 10000:20000 -j ACCEPT

iptables -A INPUT -i eth1 -p icmp --icmp-type echo-request -j DROP

iptables -A FORWARD -o eth1 -s 192.168.10.0/24 -j ACCEPT
#iptables -A FORWARD -i eth1 -d 192.168.10.0/24 -j ACCEPT
iptables -A FORWARD -m conntrack --ctstate RELATED,ESTABLISHED,DNAT -j ACCEPT
iptables -t nat -A POSTROUTING -o eth1 -s 192.168.10.0/24 -j MASQUERADE

iptables -A INPUT -i eth1 -j DROP
Поясню - машина с двумя интерфейсами (eth0 локалка и eth1 с белым IP). Инет перекидываю маскарадом, как Вы можете видеть. Возможно это не совсем правильное решение и поэтому у меня глюки? Но ведь другие провы работают. Проблемный провайдер 195.47.255.119. Из дебага я увидел, что он еще инвайтит IP 172.16.240.20. Когда я подключался к этом прову они мне сказали, что 195.47.255.119 внешний IP (типа их NAT O_O), а 172.16.240.20 их внутренний IP уже сервака. Поэтому я прописал в фаере правило и для этого IP.
В дебаге есть инфа, что я Cisco, немного поясню - без строчек в sip_general_custom.conf:

Код: Выделить всё

useragent=Cisco-SIPGateway/IOS-12.x
sdpsession=SIP Call
sdpowner=Cisco-SIPUA
вообще никакой голос не ходит.
Настройки пира:

Код: Выделить всё

host=195.47.255.119
type=peer
disallow=all
allow=alaw&g729
dtmfmode=rfc2833
nat=no
qualify=yes
canreinvite=no
insecure=port&invite
context=from-trunk
По маршрутизации все очень просто - при попадании звонка на номер этого провайдера кидаю в единую группу.
Установлен дистрибутив FreePBX 5.211.65-13; Asterisk 11.9.0.
Я уже и не знаю, в чем проблема. По идее это либо NAT либо кодеки, но пробовал играться с ними - изменений нет =(

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 14:36
ded
Yazov писал(а): Проблемный провайдер 195.47.255.119. Из дебага я увидел, что он еще инвайтит IP 172.16.240.20. Когда я подключался к этом прову они мне сказали, что 195.47.255.119 внешний IP (типа их NAT O_O), а 172.16.240.20 их внутренний IP уже сервака.
nat=yes
Изображение

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 14:55
Yazov
Прошу прощения. Всегда стоял nat=yes, это я эксперементировал. Вернул сейчас в nat=yes. Пробовал так же nat=force_rport,comedia
Безрезультатно.
RTP бегают в обе стороны.

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 18:19
alex286
m=audio 46160 RTP/AVP 8 97 ну и как через это порт "будет говорить"?

P.S. Пусть старшие товарищи поправят.. может и не прав..

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 18:37
Vlad1983
неправ

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 18:49
Vlad1983
так и не понял из всего скомканного описания астериск за NAT или нет?

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 18:58
ded
Там (якобы) у провайдера НАТ, и у ТС тоже НАТ.
Со всеми другими провами ТС коммуницируется нормально, а один - проблемный.

Re: Нет звука при входящем звонке

Добавлено: 26 июн 2014, 20:57
Vlad1983
Вырисовалась таже картинка, но что скажет ТС?

Re: Нет звука при входящем звонке

Добавлено: 27 июн 2014, 08:06
Yazov
Vlad1983 писал(а):так и не понял из всего скомканного описания астериск за NAT или нет?
Моя машина не спрятана за NAT. У меня канал прямо в eth1 идет. Какой уж тут нат? А вот у првайдера у NAT имеется, судя по всему.
Нашел такое:
<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK5a23a27a;rport=5060
From: "Unknown" <sip:Unknown@195.189.29.18>;tag=as6349f64e
To: <sip:195.47.255.119>;tag=aprqngfrt-29bp0220000c6
Call-ID: 0192ccfc1ab91ac668466660545ae6c5@195.189.29.18:5060
CSeq: 102 OPTIONS
Выставил Force Trunk CID - без изменений.

Вытащил полный дебаг проблемного звонка (как раз когда не слышно):
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Debug

Код: Выделить всё

SIP Debugging Enabled for IP: 195.47.255.119
Reliably Transmitting (NAT) to 195.47.255.119:5060:
OPTIONS sip:195.47.255.119 SIP/2.0
Via: SIP/2.0/UDP 195.189.29.18:5060;branch=z9hG4bK7b34ea4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@195.189.29.18>;tag=as365e0676
To: <sip:195.47.255.119>
Contact: <sip:Unknown@195.189.29.18:5060>
Call-ID: 694722a34a3fd61e7c9b420d24744a54@195.189.29.18:5060
CSeq: 102 OPTIONS
User-Agent: Cisco-SIPGateway/IOS-12.x
Date: Fri, 27 Jun 2014 04:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK7b34ea4f;rport=5060
From: "Unknown" <sip:Unknown@195.189.29.18>;tag=as365e0676
To: <sip:195.47.255.119>;tag=aprqngfrt-84nd0f10000c6
Call-ID: 694722a34a3fd61e7c9b420d24744a54@195.189.29.18:5060
CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '694722a34a3fd61e7c9b420d24744a54@195.189.29.18:5060' Method: OPTIONS
[2014-06-27 10:24:31] WARNING[5513]: chan_sip.c:4176 retrans_pkt: Retransmission timeout reached on transmission 75b8d21254789c242ddec59e4c46e1cf@192.168.100.227:5060 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response

<--- SIP read from UDP:195.47.255.119:5060 --->
INVITE sip:+77787460016@195.189.29.18:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK0i6ld120588h6tkao420.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>
Max-Forwards: 69
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4353 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 434

v=0
o=- 14312820 14312820 IN IP4 195.47.255.119
s=-
c=IN IP4 195.47.255.119
t=0 0
a=sendrecv
m=audio 24882 RTP/AVP 8 96 18 97
c=IN IP4 195.47.255.119
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (13 headers 18 lines) ---
Sending to 195.47.255.119:5060 (no NAT)
Sending to 195.47.255.119:5060 (no NAT)
Using INVITE request as basis request - SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
Found peer 'Kcell2' for '7078121170' from 195.47.255.119:5060
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 18
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found unknown media description format AMR for ID 96
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 195.47.255.119:24882
Looking for +77787460016 in from-trunk (domain 195.189.29.18)
list_route: hop: <sip:172.16.240.20:5060;transport=udp>

<--- Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK0i6ld120588h6tkao420.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4353 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Length: 0


<------------>
Audio is at 19180
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK0i6ld120588h6tkao420.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4353 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Type: application/sdp
Content-Length: 275

v=0
o=Cisco-SIPUA 927843942 927843942 IN IP4 195.189.29.18
s=SIP Call
c=IN IP4 195.189.29.18
t=0 0
m=audio 19180 RTP/AVP 8 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:195.47.255.119:5060 --->
ACK sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Max-Forwards: 69
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4353 ACK
Content-Length: 0

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

<--- SIP read from UDP:195.47.255.119:5060 --->
INVITE sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0cb0002110.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Max-Forwards: 69
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4354 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 239

v=0
o=- 14312820 14312821 IN IP4 195.47.255.119
s=-
c=IN IP4 195.47.255.119
t=0 0
a=sendrecv
m=audio 24882 RTP/AVP 8 97
c=IN IP4 195.47.255.119
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
--- (12 headers 12 lines) ---
Sending to 195.47.255.119:5060 (NAT)
Found RTP audio format 8
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 195.47.255.119:24882

<--- Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0cb0002110.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4354 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Length: 0


<------------>
Audio is at 19180
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0cb0002110.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4354 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+77787460016@195.189.29.18:5060>
Content-Type: application/sdp
Content-Length: 228

v=0
o=Cisco-SIPUA 927843942 927843943 IN IP4 195.189.29.18
s=SIP Call
c=IN IP4 195.189.29.18
t=0 0
m=audio 19180 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:195.47.255.119:5060 --->
ACK sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK8irg8m107ojghsokn411.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Max-Forwards: 69
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4354 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[2014-06-27 10:24:34] WARNING[6743][C-00000068]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443303
Audio is at 16682
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.47.255.119:5060:
INVITE sip:+77051707960@195.47.255.119 SIP/2.0
Via: SIP/2.0/UDP 195.189.29.18:5060;branch=z9hG4bK61e7c447;rport
Max-Forwards: 70
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>
Contact: <sip:+77787460016@195.189.29.18:5060>
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 INVITE
User-Agent: Cisco-SIPGateway/IOS-12.x
Date: Fri, 27 Jun 2014 04:24:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=Cisco-SIPUA 1103547086 1103547086 IN IP4 195.189.29.18
s=SIP Call
c=IN IP4 195.189.29.18
t=0 0
m=audio 16682 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK61e7c447;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 INVITE

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

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK61e7c447;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>;tag=SD0njr599-0408515094
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 242

v=0
o=- 10521094 10521094 IN IP4 195.47.255.119
s=-
c=IN IP4 195.47.255.119
t=0 0
a=sendrecv
m=audio 23484 RTP/AVP 8 101
c=IN IP4 195.47.255.119
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:40
<------------->
--- (10 headers 12 lines) ---
list_route: hop: <sip:172.16.240.20:5060;transport=udp>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 195.47.255.119:23484

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK61e7c447;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>;tag=SD0njr599-0408515094
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 INVITE
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: hop: <sip:172.16.240.20:5060;transport=udp>
Scheduling destruction of SIP dialog '1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 195.47.255.119:5060:
CANCEL sip:+77051707960@195.47.255.119 SIP/2.0
Via: SIP/2.0/UDP 195.189.29.18:5060;branch=z9hG4bK61e7c447;rport
Max-Forwards: 70
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 CANCEL
User-Agent: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


---
Scheduling destruction of SIP dialog '1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK61e7c447;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>;tag=SD0njr599-0408515094
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 CANCEL

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

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK61e7c447;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>;tag=SD0njr599-0408515094
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 INVITE
Reason: Q.850;cause=31
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 195.47.255.119:5060:
ACK sip:172.16.240.20:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.189.29.18:5060;branch=z9hG4bK61e7c447;rport
Max-Forwards: 70
From: <sip:+77787460016@195.189.29.18>;tag=as67c9080d
To: <sip:+77051707960@195.47.255.119>;tag=SD0njr599-0408515094
Contact: <sip:+77787460016@195.189.29.18:5060>
Call-ID: 1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060
CSeq: 102 ACK
User-Agent: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


---
Scheduling destruction of SIP dialog '1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060' in 6400 ms (Method: INVITE)
[2014-06-27 10:24:44] WARNING[6743][C-00000068]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 54414700
[2014-06-27 10:24:44] WARNING[6743][C-00000068]: mp3/common.c:135 decode_header: Layer 1 not supported!
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-06-27 10:24:45] WARNING[6743][C-00000068]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Really destroying SIP dialog '1ebbb6ab5eebe859535e6bd50e258313@195.189.29.18:5060' Method: INVITE
Scheduling destruction of SIP dialog '5b3008b0608d907f1a6476151f162491@195.189.29.18:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:172.16.240.20:5060;transport=udp> for address/port to send to
set_destination: set destination to 172.16.240.20:5060
Reliably Transmitting (NAT) to 195.47.255.119:5060:
BYE sip:172.16.240.20:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.189.29.18:5060;branch=z9hG4bK5fec2583;rport
Max-Forwards: 70
From: <sip:+77787460016@195.189.29.18>;tag=as22694576
To: <sip:+77778560070@195.47.255.119>;tag=SD6ngnd99-0061728853
Call-ID: 5b3008b0608d907f1a6476151f162491@195.189.29.18:5060
CSeq: 103 BYE
User-Agent: Cisco-SIPGateway/IOS-12.x
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.189.29.18:5060;received=195.189.29.18;branch=z9hG4bK5fec2583;rport=5060
From: <sip:+77787460016@195.189.29.18>;tag=as22694576
To: <sip:+77778560070@195.47.255.119>;tag=SD6ngnd99-0061728853
Call-ID: 5b3008b0608d907f1a6476151f162491@195.189.29.18:5060
CSeq: 103 BYE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5b3008b0608d907f1a6476151f162491@195.189.29.18:5060' Method: INVITE

<--- SIP read from UDP:195.47.255.119:5060 --->
BYE sip:+77787460016@195.189.29.18:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0cd0002h10.1
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Max-Forwards: 69
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4355 BYE
Reason: Q.850;cause=16
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 195.47.255.119:5060 (NAT)
Scheduling destruction of SIP dialog 'SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 195.47.255.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.240.20:5060;branch=z9hG4bK6i180i007010evgc13k0cd0002h10.1;received=195.47.255.119;rport=5060
From: <sip:7078121170@172.16.240.20:5060;user=phone>;tag=SD9ned301-2299058074
To: <sip:+77787460016@195.189.29.18:5060;user=phone>;tag=as708bf540
Call-ID: SD9ned301-9b3818e2a95d7a02798d4e9fbe6e70c6-v300g00
CSeq: 4355 BYE
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Re: Нет звука при входящем звонке

Добавлено: 27 июн 2014, 08:26
Vlad1983
выставить на пире этого оператора

Код: Выделить всё

outboundproxy=195.47.255.119
nat=yes
Часовой пояс: UTC+03:00
Страница 1 из 3

Создано на основе phpBB® Forum Software © phpBB Limited

Русская поддержка phpBB