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Звонки идут в обе стороны, голос не идет

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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dmitriy.tarasov
Сообщения: 17
Зарегистрирован: 07 авг 2013, 17:03

Звонки идут в обе стороны, голос не идет

Сообщение dmitriy.tarasov »

Мир всем.
Возникла задача соединить несколько удаленных офисов. Офисы соединены по VPN. Схема следующая:

Digium D40 <-> Firewall <-> VPN <-> FRW <-> Switch <-> Asterisk SRV
.................................................................................<-> Kerio <-> Digium D40

Звонки пошли в обе стороны без проблем. голос не ходит ни одну сторону. NAT-а нет, трафик просто рутится в обе стороны.
Где можно смотреть в чем проблема

PS:
directmedia=yes
qualify=yes
rtp.conf 10000-20000
на фаерволе открыты все порты к телефонам и к серверу Астериск в обе стороны
Последний раз редактировалось dmitriy.tarasov 07 авг 2013, 17:36, всего редактировалось 3 раза.
Вернуться к началу
ded
Сообщения: 15620
Зарегистрирован: 26 авг 2010, 19:00

Re: Звонки идут в обе стороны, голос не идет

Сообщение ded »

Загляните в раздел для новичков. Там две блок схемы, отвечают на вопрос "Куда смотреть?"
Вернуться к началу
jugatsu
Сообщения: 298
Зарегистрирован: 31 май 2011, 15:56

Re: Звонки идут в обе стороны, голос не идет

Сообщение jugatsu »

Дамп, сдп ну и т.д.
Вернуться к началу
dmitriy.tarasov
Сообщения: 17
Зарегистрирован: 07 авг 2013, 17:03

Re: Звонки идут в обе стороны, голос не идет

Сообщение dmitriy.tarasov »

ded, спасибо, пойду искать эти схемы
jugatsu. Вот сип дебаг:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
[Aug 7 16:18:07] Asterisk 1.8.11-cert7, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
[Aug 7 16:18:07] Connected to Asterisk 1.8.11-cert7 currently running on localhost (pid = 27247)
localhost*CLI> sip set debug 1778
localhost*CLI>
[0KNo such command 'sip set debug 1778' (type 'core show help sip set debug' for other possible commands)

[Klocalhost*CLI> sip set debug 1778[K[K[K[K
ip
off
on
peer

[Klocalhost*CLI> sip set debug peer 1778
localhost*CLI>
[0KSIP Debugging Enabled for IP: 172.16.3.40

[Klocalhost*CLI> sip set debug peer 1778[K[K[K[K5142
localhost*CLI>
[0KSIP Debugging Enabled for IP: 172.16.112.6

[Klocalhost*CLI>
[0KReally destroying SIP dialog 'Tq19eFS1XcTNRj9bhiHkH-pCU4s5yiYp' Method: REGISTER

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
INVITE sip:1778@172.16.0.70 SIP/2.0
Via: SIP/2.0/UDP 172.16.112.6:5060;rport;branch=z9hG4bKPjaVKehzJya2wjLgIEUp.ncmycghU.KzH8
Max-Forwards: 70
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10580 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Digium D40 1_3_2_0_54993
Content-Type: application/sdp
Content-Length: 431

v=0
o=- 71245624 71245624 IN IP4 172.16.112.6
s=digphn
c=IN IP4 172.16.112.6
t=0 0
a=X-nat:0
m=audio 4038 RTP/AVP 0 8 9 111 18 58 118 58 96
a=rtcp:4039 IN IP4 172.16.112.6
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:58 L16/16000
a=rtpmap:118 L16/8000
a=rtpmap:58 L16-256/16000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
--- (15 headers 19 lines) ---
Sending to 172.16.112.6:5060 (NAT)
Using INVITE request as basis request - 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
Found peer '5142' for '5142' from 172.16.112.6:5060

[Klocalhost*CLI>
[0K
<--- Reliably Transmitting (NAT) to 172.16.112.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjaVKehzJya2wjLgIEUp.ncmycghU.KzH8;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as7b09e62a
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10580 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="233d388d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY' in 6400 ms (Method: INVITE)

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
ACK sip:1778@172.16.0.70 SIP/2.0
Via: SIP/2.0/UDP 172.16.112.6:5060;rport;branch=z9hG4bKPjaVKehzJya2wjLgIEUp.ncmycghU.KzH8
Max-Forwards: 70
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as7b09e62a
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10580 ACK
Content-Length: 0

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

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
INVITE sip:1778@172.16.0.70 SIP/2.0
Via: SIP/2.0/UDP 172.16.112.6:5060;rport;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH
Max-Forwards: 70
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Digium D40 1_3_2_0_54993
Authorization: Digest username="5142", realm="asterisk", nonce="233d388d", uri="sip:1778@172.16.0.70", response="321735823e6399a0757995fc7f1a1ad6", algorithm=MD5
Content-Type: application/sdp
Content-Length: 431

v=0
o=- 71245624 71245624 IN IP4 172.16.112.6
s=digphn
c=IN IP4 172.16.112.6
t=0 0
a=X-nat:0
m=audio 4038 RTP/AVP 0 8 9 111 18 58 118 58 96
a=rtcp:4039 IN IP4 172.16.112.6
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:58 L16/16000
a=rtpmap:118 L16/8000
a=rtpmap:58 L16-256/16000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
--- (16 headers 19 lines) ---
Sending to 172.16.112.6:5060 (NAT)
Using INVITE request as basis request - 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
Found peer '5142' for '5142' from 172.16.112.6:5060

[Klocalhost*CLI>
[0KFound RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 111
Found RTP audio format 18
Found RTP audio format 58
Found RTP audio format 118
Found RTP audio format 58
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 111
Found audio description format G729 for ID 18
Found audio description format L16 for ID 58
Found audio description format L16 for ID 118
Found audio description format L16-256 for ID 58
Found audio description format telephone-event for ID 96
Capabilities: us - 0x180c (ulaw|alaw|g726|g722), peer - audio=0x994c (ulaw|alaw|g726|slin|g729|g722|slin16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x180c (ulaw|alaw|g726|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.112.6:4038
Looking for 1778 in M (domain 172.16.0.70)

[Klocalhost*CLI>
[0Klist_route: hop: <sip:5142@172.16.112.6:5060;ob>

[Klocalhost*CLI>
[0Kset_destination: Parsing <sip:5142@172.16.112.6:5060;ob> for address/port to send to

<--- Transmitting (NAT) to 172.16.112.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Length: 0


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

[Klocalhost*CLI>
[0Kset_destination: set destination to 172.16.112.6:5060
Reliably Transmitting (NAT) to 172.16.112.6:5060:
NOTIFY sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK11fcfc77;rport
Max-Forwards: 70
From: sip:auto_hint_5142@172.16.0.70;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
Contact: <sip:auto_hint_5142@172.16.0.70:5060>
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX 1.8.11-cert7
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 667

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:5142@172.16.0.70">
<pp:person><status>
<ep:activities><ep:busy/></ep:activities>
</status></pp:person>
<note>On the phone</note>
<tuple id="auto_hint_5142">
<contact priority="1">sip:auto_hint_5142@172.16.0.70</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK11fcfc77
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
From: <sip:auto_hint_5142@172.16.0.70>;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
CSeq: 107 NOTIFY
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

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

[Klocalhost*CLI>
[0K
<--- Transmitting (NAT) to 172.16.112.6:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Length: 0


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

[Klocalhost*CLI>
[0KAudio is at 11824
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8 (alaw) to SDP

[Klocalhost*CLI>
[0KAdding codec 0x4 (ulaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

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

[Klocalhost*CLI>
[0KRetransmitting #1 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KRetransmitting #2 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KRetransmitting #3 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KRetransmitting #4 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KReliably Transmitting (NAT) to 172.16.112.6:5060:
OPTIONS sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK128159ea;rport
Max-Forwards: 70
From: "kanardan-zang" <sip:kanardan-zang@172.16.0.70>;tag=as64a3d5f4
To: <sip:5142@172.16.112.6:5060;ob>
Contact: <sip:kanardan-zang@172.16.0.70:5060>
Call-ID: 69666b4c59bd628d091bbe9b74f0962a@172.16.0.70:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert7
Date: Wed, 07 Aug 2013 11:18:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK128159ea
Call-ID: 69666b4c59bd628d091bbe9b74f0962a@172.16.0.70:5060
From: "kanardan-zang" <sip:kanardan-zang@172.16.0.70>;tag=as64a3d5f4
To: <sip:5142@172.16.112.6;ob>;tag=z9hG4bK128159ea
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Digium D40 1_3_2_0_54993
Content-Type: application/sdp
Content-Length: 420

v=0
o=- 71245628 71245628 IN IP4 172.16.112.6
s=digphn
c=IN IP4 172.16.112.6
t=0 0
m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96
a=rtcp:4001 IN IP4 172.16.112.6
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:58 L16/16000
a=rtpmap:118 L16/8000
a=rtpmap:58 L16-256/16000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
--- (13 headers 18 lines) ---
Really destroying SIP dialog '69666b4c59bd628d091bbe9b74f0962a@172.16.0.70:5060' Method: OPTIONS

[Klocalhost*CLI>
[0KRetransmitting #5 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KScheduling destruction of SIP dialog '9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY' in 6400 ms (Method: INVITE)

[Klocalhost*CLI>
[0Kset_destination: Parsing <sip:5142@172.16.112.6:5060;ob> for address/port to send to

[Klocalhost*CLI>
[0Kset_destination: set destination to 172.16.112.6:5060

[Klocalhost*CLI>
[0KReliably Transmitting (NAT) to 172.16.112.6:5060:
NOTIFY sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK711a065f;rport
Max-Forwards: 70
From: sip:auto_hint_5142@172.16.0.70;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
Contact: <sip:auto_hint_5142@172.16.0.70:5060>
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX 1.8.11-cert7
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 618

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:5142@172.16.0.70">
<pp:person><status>
</status></pp:person>
<note>Ready</note>
<tuple id="auto_hint_5142">
<contact priority="1">sip:auto_hint_5142@172.16.0.70</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK711a065f
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
From: <sip:auto_hint_5142@172.16.0.70>;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
CSeq: 108 NOTIFY
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

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

[Klocalhost*CLI>
[0KRetransmitting #6 (NAT) to 172.16.112.6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as624ea483
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 INVITE
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1778@172.16.0.70:5060>
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1185857321 1185857321 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 11824 RTP/AVP 9 8 0 111 96
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0KReally destroying SIP dialog '9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY' Method: INVITE

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
CANCEL sip:1778@172.16.0.70 SIP/2.0
Via: SIP/2.0/UDP 172.16.112.6:5060;rport;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH
Max-Forwards: 70
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 CANCEL
User-Agent: Digium D40 1_3_2_0_54993
Content-Length: 0

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

<--- Transmitting (NAT) to 172.16.112.6:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 172.16.112.6:5060;branch=z9hG4bKPjby3Yu7W9sb3pK.mtvE8bBC19aHlf0seH;received=172.16.112.6;rport=5060
From: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=QiZMbiabDWbKO5HEYWqzwgTZvfK8N7UT
To: <sip:1778@172.16.0.70>;tag=as07f2d36a
Call-ID: 9wP2mzpXEczG6xMH78lXtjfM5SpM9.PY
CSeq: 10581 CANCEL
Server: Asterisk PBX 1.8.11-cert7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

[Klocalhost*CLI>
[0KAudio is at 17674
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.16.112.6:5060:
INVITE sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK6b082ea7;rport
Max-Forwards: 70
From: "Şәbәkә 02" <sip:1778@172.16.0.70>;tag=as14c4747e
To: <sip:5142@172.16.112.6:5060;ob>
Contact: <sip:1778@172.16.0.70:5060>
Call-ID: 675117044431463539302e870ec1578a@172.16.0.70:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11-cert7
Date: Wed, 07 Aug 2013 11:19:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 1727630951 1727630951 IN IP4 172.16.0.70
s=Asterisk PBX 1.8.11-cert7
c=IN IP4 172.16.0.70
t=0 0
m=audio 17674 RTP/AVP 9 8 0 111 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK6b082ea7
Call-ID: 675117044431463539302e870ec1578a@172.16.0.70:5060
From: "Şәbәkә 02" <sip:1778@172.16.0.70>;tag=as14c4747e
To: <sip:5142@172.16.112.6;ob>
CSeq: 102 INVITE
Content-Length: 0

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

[Klocalhost*CLI>
[0Kset_destination: Parsing <sip:5142@172.16.112.6:5060;ob> for address/port to send to

[Klocalhost*CLI>
[0Kset_destination: set destination to 172.16.112.6:5060

[Klocalhost*CLI>
[0KReliably Transmitting (NAT) to 172.16.112.6:5060:
NOTIFY sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK1d1db0ba;rport
Max-Forwards: 70
From: sip:auto_hint_5142@172.16.0.70;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
Contact: <sip:auto_hint_5142@172.16.0.70:5060>
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
CSeq: 109 NOTIFY
User-Agent: Asterisk PBX 1.8.11-cert7
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 662

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:5142@172.16.0.70">
<pp:person><status>
<ep:activities><ep:busy/></ep:activities>
</status></pp:person>
<note>Ringing</note>
<tuple id="auto_hint_5142">
<contact priority="1">sip:auto_hint_5142@172.16.0.70</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK1d1db0ba
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
From: <sip:auto_hint_5142@172.16.0.70>;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
CSeq: 109 NOTIFY
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

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

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK6b082ea7
Call-ID: 675117044431463539302e870ec1578a@172.16.0.70:5060
From: "Şәbәkә 02" <sip:1778@172.16.0.70>;tag=as14c4747e
To: <sip:5142@172.16.112.6;ob>;tag=uKXOYNZgFk-NoUhtlRT7P79DSq2U3PH1
CSeq: 102 INVITE
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
User-Agent: Digium D40 1_3_2_0_54993
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:5142@172.16.112.6:5060;ob>

[Klocalhost*CLI>
[0KScheduling destruction of SIP dialog '675117044431463539302e870ec1578a@172.16.0.70:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:5142@172.16.112.6:5060;ob> for address/port to send to

[Klocalhost*CLI>
[0Kset_destination: set destination to 172.16.112.6:5060
Reliably Transmitting (NAT) to 172.16.112.6:5060:
CANCEL sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK6b082ea7;rport
Max-Forwards: 70
From: "Şәbәkә 02" <sip:1778@172.16.0.70>;tag=as14c4747e
To: <sip:5142@172.16.112.6:5060;ob>
Call-ID: 675117044431463539302e870ec1578a@172.16.0.70:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.11-cert7
Content-Length: 0


---
Scheduling destruction of SIP dialog '675117044431463539302e870ec1578a@172.16.0.70:5060' in 6400 ms (Method: INVITE)

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK6b082ea7
Call-ID: 675117044431463539302e870ec1578a@172.16.0.70:5060
From: "Şәbәkә 02" <sip:1778@172.16.0.70>;tag=as14c4747e
To: <sip:5142@172.16.112.6;ob>;tag=uKXOYNZgFk-NoUhtlRT7P79DSq2U3PH1
CSeq: 102 CANCEL
Content-Length: 0

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

[Klocalhost*CLI>
[0Kset_destination: Parsing <sip:5142@172.16.112.6:5060;ob> for address/port to send to
set_destination: set destination to 172.16.112.6:5060
Reliably Transmitting (NAT) to 172.16.112.6:5060:
NOTIFY sip:5142@172.16.112.6:5060;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.0.70:5060;branch=z9hG4bK77fc6de6;rport
Max-Forwards: 70
From: sip:auto_hint_5142@172.16.0.70;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
Contact: <sip:auto_hint_5142@172.16.0.70:5060>
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX 1.8.11-cert7
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 618

<?xml version="1.0" encoding="ISO-8859-1"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:pp="urn:ietf:params:xml:ns:pidf:person"
xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"
xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"
entity="sip:5142@172.16.0.70">
<pp:person><status>
</status></pp:person>
<note>Ready</note>
<tuple id="auto_hint_5142">
<contact priority="1">sip:auto_hint_5142@172.16.0.70</contact>
<status><basic>open</basic></status>
</tuple>
<tuple id="digium-presence">
<status>
<digium_presence type="available" subtype=""></digium_presence>
</status>
</tuple>
</presence>

---

[Klocalhost*CLI>
[0K
<--- SIP read from UDP:172.16.112.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.70:5060;rport=5060;received=172.16.0.70;branch=z9hG4bK77fc6de6
Call-ID: dQRdfBkO9DvwsmVbiT9vW.Z.jHwKihnr
From: <sip:auto_hint_5142@172.16.0.70>;tag=as70f17eb9
To: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.0.70>;tag=-DS4OKfOVog34-05xEuwgEkm2IVqSe0y
CSeq: 110 NOTIFY
Contact: "\"Yas-C.Rasimә\" <5142>" <sip:5142@172.16.112.6:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

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

[Klocalhost*CLI> sip set debug peer 5142[K[K[K[K[K[K[K[Koff
localhost*CLI>
[0KSIP Debugging Disabled

[Klocalhost*CLI> exit
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Звонки идут в обе стороны, голос не идет

Сообщение awsswa »

это дебаг, когда звонков нет.
а нужен когда голоса нету - tcpdump
платный суппорт по мере возможностей
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dmitriy.tarasov
Сообщения: 17
Зарегистрирован: 07 авг 2013, 17:03

Re: Звонки идут в обе стороны, голос не идет

Сообщение dmitriy.tarasov »

Этот дебаг я снял совершив звонок с 5142 на 1778, а затем с 1778 на 5142:
core set verbose 0
core set debug 0
rtp set debug off
sip set debug peer 1778
sip set debug peer 5142
Последний раз редактировалось dmitriy.tarasov 07 авг 2013, 20:30, всего редактировалось 1 раз.
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Звонки идут в обе стороны, голос не идет

Сообщение Vlad1983 »

directmedia=no
ЛС: @rostel
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dmitriy.tarasov
Сообщения: 17
Зарегистрирован: 07 авг 2013, 17:03

Re: Звонки идут в обе стороны, голос не идет

Сообщение dmitriy.tarasov »

Я могу ошибаться, но как я понимаю, directmedia=no используется для отключения ре-инвайтов в случае, если nat=yes плюс это приведет к тому, что весь RTP трафик будет проходить не напрямую, а через сервер Asterisk. У меня NAT-а нет, плюс, если directmedia=no даже и будет работать если не используется NAT, то не создаст ли это большой нагрузки на сервер Asterisk (планируется использование около 400 хард-фонов)
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awsswa
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Откуда: Россия, Пермь skype: yarick_perm

Re: Звонки идут в обе стороны, голос не идет

Сообщение awsswa »

господи, кто таким в руки дает управление 400 телефонами ?
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ded
Сообщения: 15620
Зарегистрирован: 26 авг 2010, 19:00

Re: Звонки идут в обе стороны, голос не идет

Сообщение ded »

Дмитрий, Вы можете ошибаться.
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