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Username Secret Accountcode Def.Context ACL NAT
pelu default No RFC3581

Вводная: когда pelu есть в списке пиров - звонок проходит.
Интересно увидеть Unable create channel SIP to pelu - вот в этот момент тоже есть?

вроде Unable create channel SIP to pelu больше не появляется.

Но вот косяк, при некоторых звонках голос не проходит, косяк наблюдается например при звонке с Ekiga на Twinkle (Ekiga-Ekiga норм, Twinkle-Twinkle тоже, Twinkle-Ekiga опять норма, Ekiga-Twinkle тишина).

и все равно после некоторых звонков, если один завершает, 2й может не получить сигнал завершения разговора и оставаться busy до реконнекта :(

Вводная:
CLI>sip set debug peer Twinkle

asterisk*CLI> sip set debug peer lucia
SIP Debugging Enabled for IP: 195.209.66.96:5060
asterisk*CLI> sip set debug peer pelu
SIP Debugging Enabled for IP: 195.209.66.102:5060
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
> Saved useragent "Twinkle/1.4.2" for peer lucia
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->
INVITE sip:lucia@195.209.66.18 SIP/2.0
Date: Wed, 16 Jun 2010 14:38:42 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKee556e87-c277-df11-93a9-0026180a0f91;rport
User-Agent: Ekiga/3.2.6
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
To: <sip:lucia@195.209.66.18>
Contact: <sip:pelu@195.209.66.102>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 556
Max-Forwards: 70

v=0
o=- 1276699122 1 IN IP4 195.209.66.102
s=Opal SIP Session
c=IN IP4 195.209.66.102
t=0 0
m=audio 5078 RTP/AVP 114 0 8 9 101 120
a=sendrecv
a=rtpmap:114 Speex/16000/1
a=fmtp:114 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5080 RTP/AVP 119 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:119 theora/90000
a=fmtp:119 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1

<------------->
--- (13 headers 24 lines) ---
== Using SIP RTP CoS mark 5
Sending to 195.209.66.102 : 5060 (no NAT)
Using INVITE request as basis request - 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
Found peer 'pelu' for 'pelu' from 195.209.66.102:5060
Found RTP audio format 114
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found RTP audio format 120
Found audio description format Speex for ID 114
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 telephone-event for ID 101
Found audio description format NSE for ID 120
Found RTP video format 119
Found RTP video format 31
Found video description format h261 for ID 31
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x100c (ulaw|alaw|g722)/video=0x40000 (h261)/text=0x0 (nothing), combined - 0x4100c (ulaw|alaw|g722|h261)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 195.209.66.102:5078
Looking for lucia in default (domain 195.209.66.18)
list_route: hop: <sip:pelu@195.209.66.102>

<--- Transmitting (no NAT) to 195.209.66.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKee556e87-c277-df11-93a9-0026180a0f91;received=195.209.66.102;rport=5060
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
To: <sip:lucia@195.209.66.18>
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:lucia@195.209.66.18>
Content-Length: 0


<------------>
-- Executing [lucia@default:1] Macro("SIP/pelu-000000ad", "stdext,lucia") in new stack
-- Executing [s@macro-stdext:1] Dial("SIP/pelu-000000ad", "SIP/lucia,25") in new stack
== Using SIP RTP CoS mark 5
-- Called lucia
-- SIP/lucia-000000ae is ringing

<--- Transmitting (no NAT) to 195.209.66.102:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKee556e87-c277-df11-93a9-0026180a0f91;received=195.209.66.102;rport=5060
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
To: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:lucia@195.209.66.18>
Content-Length: 0


<------------>
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
-- SIP/lucia-000000ae answered SIP/pelu-000000ad
Audio is at 195.209.66.18 port 18554
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 195.209.66.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKee556e87-c277-df11-93a9-0026180a0f91;received=195.209.66.102;rport=5060
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
To: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:lucia@195.209.66.18>
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1101842075 1101842075 IN IP4 195.209.66.18
s=Asterisk PBX 1.6.2.8
c=IN IP4 195.209.66.18
t=0 0
m=audio 18554 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 119 31

<------------>
-- Native bridging SIP/pelu-000000ad and SIP/lucia-000000ae
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->
ACK sip:lucia@195.209.66.18 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bK34b9768f-c277-df11-93a9-0026180a0f91;rport
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
To: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Contact: <sip:pelu@195.209.66.102>
Content-Length: 0
Max-Forwards: 70


<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:pelu@195.209.66.102> for address/port to send to
set_destination: set destination to 195.209.66.102, port 5060
Audio is at 195.209.66.18 port 18554
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.209.66.102:5060:
INVITE sip:pelu@195.209.66.102 SIP/2.0
Via: SIP/2.0/UDP 195.209.66.18:5060;branch=z9hG4bK7b9ff0d7;rport
Max-Forwards: 70
From: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
To: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Contact: <sip:lucia@195.209.66.18>
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 1101842075 1101842076 IN IP4 195.209.66.96
s=Asterisk PBX 1.6.2.8
c=IN IP4 195.209.66.96
t=0 0
m=audio 8000 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 195.209.66.18:5060;branch=z9hG4bK7b9ff0d7;rport
From: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
To: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Contact: <sip:pelu@195.209.66.102>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 195.209.66.18:5060;branch=z9hG4bK7b9ff0d7;rport
User-Agent: Ekiga/3.2.6
From: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
To: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Contact: <sip:pelu@195.209.66.102>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 219

v=0
o=- 1276699122 2 IN IP4 195.209.66.102
s=Opal SIP Session
c=IN IP4 195.209.66.102
t=0 0
m=audio 5078 RTP/AVP 9 101
a=sendrecv
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36

<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 9
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 195.209.66.102:5078
set_destination: Parsing <sip:pelu@195.209.66.102> for address/port to send to
set_destination: set destination to 195.209.66.102, port 5060
Transmitting (no NAT) to 195.209.66.102:5060:
ACK sip:pelu@195.209.66.102 SIP/2.0
Via: SIP/2.0/UDP 195.209.66.18:5060;branch=z9hG4bK6a5a8d81;rport
Max-Forwards: 70
From: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
To: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Contact: <sip:lucia@195.209.66.18>
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.8
Content-Length: 0


---
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->


<------------->
asterisk*CLI>
<--- SIP read from UDP:195.209.66.102:5060 --->
BYE sip:lucia@195.209.66.18 SIP/2.0
CSeq: 2 BYE
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKd2fb1c95-c277-df11-93a9-0026180a0f91;rport
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
To: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Contact: <sip:pelu@195.209.66.102>
Content-Length: 0
Max-Forwards: 70


<------------->
--- (9 headers 0 lines) ---
Sending to 195.209.66.102 : 5060 (no NAT)

<--- Transmitting (no NAT) to 195.209.66.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.209.66.102:5060;branch=z9hG4bKd2fb1c95-c277-df11-93a9-0026180a0f91;received=195.209.66.102;rport=5060
From: <sip:pelu@195.209.66.18>;tag=64a96d87-c277-df11-93a9-0026180a0f91
To: <sip:lucia@195.209.66.18>;tag=as3b83a7e6
Call-ID: 8eac6d87-c277-df11-93a9-0026180a0f91@nibiru
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (macro-stdext, s, 1) exited non-zero on 'SIP/pelu-000000ad' in macro 'stdext'
== Spawn extension (default, lucia, 1) exited non-zero on 'SIP/pelu-000000ad'
Really destroying SIP dialog '8eac6d87-c277-df11-93a9-0026180a0f91@nibiru' Method: BYE


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

m=audio 5078 RTP/AVP 9 101
a=sendrecv
a=rtpmap:9 G722/8000/1

Понятно? Или нужны комментарии?
Комментарии: уберите экзотику, оставьте просто alaw & ulaw на всех фонах.
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

Позже наверняка научитесь разбираться с преференциями кодеков. Но это завтра, не сегодня.

Спасибо огромное. Оставил только ulaw и alaw и все клиенты отлично дозваниваются, никаких busy.

Только в kphone нет звука, хотя голос проходит (это как то связано с тем, что у него в настройках из кодеков только g711 gsm iLBC?). И при звонке с QuteCom такая же фигня (обратный работает)