asterisk.ru

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

Звук при исходящем звонке появляется спустя некоторое время

https://forum.asterisk.ru/viewtopic.php?t=5527

Страница 1 из 1

Звук при исходящем звонке появляется спустя некоторое время

Добавлено: 29 янв 2015, 11:07
sm_sergey
Добрый день. Возникла следующая проблема:
При исходящем звонке (например, звоню себе на сотовый) тишина в трубке в течении примерно минуту. То есть, звонок на телефон приходит, снимаю трубку, а там тишина. А со стороны asterisk по прежнему идут гудки, как будто трубка на том конце не снята. И так в течении где то минуты, затем гудки прекращаются и всё начинает работать нормально, голос появляется
===============
CentOS release 6.4 (Final)
Asterisk 1.8
===============
Sip.conf

[general]
allowoverlap=no
srvlookup=no
bindport=5060
language=ru
udpbindaddr=0.0.0.0
tcpenable=no
canreinvite=no
context=internal
callevents=yes
allowguest=yes

[trunk]

type=friend
context=internal
host=192.168.1.250
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
allow=gsm

===============

extensions.conf

exten => _9XXXXXX,1,Dial(SIP/192.168.1.250/${EXTEN},60,tTxX)
exten => _08XXXXXXXXXX,1,Dial(SIP/192.168.1.250/${EXTEN},60,tTxX)
exten => _98XXXXXXXXXX,1,Dial(SIP/192.168.1.250/${EXTEN},60,tTxX)

===============
sip debug

-- Accepting AUTHENTICATED call from 192.168.6.12:
> requested format = gsm,
> requested prefs = (),
> actual format = alaw,
> host prefs = (g729|alaw|ulaw|gsm),
> priority = mine
-- Executing [089516922533@default:1] Dial("IAX2/Alex1-5214", "SIP/192.168.1.250/089516922533,60,tTxX") in new stack
== Using SIP RTP CoS mark 5
Audio is at 19738
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.250:5060:
INVITE sip:089516922533@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK484ab354;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Date: Thu, 29 Jan 2015 04:23:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 304

v=0
o=root 1991313055 1991313055 IN IP4 192.168.111.163
s=Asterisk PBX SVN-branch-1.8-r397377
c=IN IP4 192.168.111.163
t=0 0
m=audio 19738 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/192.168.1.250/089516922533

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 100 Trying
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
Contact: <sip:089516922533@192.168.1.250>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 183 Session Progress
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 111
Found audio description format telephone-event for ID 101
Found unknown media description format X-nt-inforeq for ID 111
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.231:5248
-- SIP/192.168.1.250-0000002c is making progress passing it to IAX2/Alex1-5214

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 180 Ringing
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
list_route: hop: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
-- SIP/192.168.1.250-0000002c is ringing
-- SIP/192.168.1.250-0000002c is making progress passing it to IAX2/Alex1-5214

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
list_route: hop: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
set_destination: Parsing <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone> for address/port to send to
[Jan 29 08:23:31] WARNING[2812]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:23:32] WARNING[2811]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:23:39] ERROR[2816]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("smolensk.sml", "5060", ...): Name or service not known
[Jan 29 08:23:39] WARNING[2816]: chan_sip.c:10583 set_destination: Can't find address for host 'smolensk.sml:5060'
Transmitting (NAT) to 192.168.1.250:5060:
ACK sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK5aab9e0f;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
set_destination: Parsing <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone> for address/port to send to
[Jan 29 08:23:40] WARNING[2808]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:23:49] ERROR[2816]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("smolensk.sml", "5060", ...): Name or service not known
[Jan 29 08:23:49] WARNING[2816]: chan_sip.c:10583 set_destination: Can't find address for host 'smolensk.sml:5060'
Transmitting (NAT) to 192.168.1.250:5060:
ACK sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK4430320a;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
set_destination: Parsing <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone> for address/port to send to
[Jan 29 08:23:50] WARNING[2813]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:23:59] ERROR[2816]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("smolensk.sml", "5060", ...): Name or service not known
[Jan 29 08:23:59] WARNING[2816]: chan_sip.c:10583 set_destination: Can't find address for host 'smolensk.sml:5060'
Transmitting (NAT) to 192.168.1.250:5060:
ACK sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK43df69ad;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
set_destination: Parsing <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone> for address/port to send to
[Jan 29 08:24:00] WARNING[2813]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:24:09] ERROR[2816]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("smolensk.sml", "5060", ...): Name or service not known
[Jan 29 08:24:09] WARNING[2816]: chan_sip.c:10583 set_destination: Can't find address for host 'smolensk.sml:5060'
Transmitting (NAT) to 192.168.1.250:5060:
ACK sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK1915ecba;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.111.163:5060;rport=5060;branch=z9hG4bK484ab354
Supported: 100rel,x-nortel-sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.00.20
P-Asserted-Identity: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml;user=phone>
Privacy: none
Contact: <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2087965 1 IN IP4 192.168.1.250
s=-
c=IN IP4 192.168.1.231
t=0 0
m=audio 5248 RTP/AVP 8 101 111
c=IN IP4 192.168.1.231
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
set_destination: Parsing <sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone> for address/port to send to
[Jan 29 08:24:09] WARNING[2805]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/Alex1-5214
[Jan 29 08:24:19] ERROR[2816]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("smolensk.sml", "5060", ...): Name or service not known
[Jan 29 08:24:19] WARNING[2816]: chan_sip.c:10583 set_destination: Can't find address for host 'smolensk.sml:5060'
Transmitting (NAT) to 192.168.1.250:5060:
ACK sip:089516922533;phone-context=UnknownUnknown@smolensk.sml:5060;maddr=192.168.1.250;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.111.163:5060;branch=z9hG4bK10e3b38a;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>;tag=78821e8-fa01a8c0-13c4-55013-14bfb34-3e4ca03-14bfb34
Contact: <sip:anonymous@192.168.111.163:5060>
Call-ID: 538ff80000eb5abf517c7dc64a568f04@192.168.111.163:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r397377
Content-Length: 0


---
-- SIP/192.168.1.250-0000002c answered IAX2/Alex1-5214

===============
Сервер стоит уже два года и подобных проблем не наблюдалось. Кто нибудь с подобным сталкивался? Спасибо.

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 11:45
ded

Код: Выделить всё

SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as354db4eb
To: <sip:089516922533@192.168.1.250>
не смущает?

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 12:03
sm_sergey
Да в общем нет. Через zoiper по iax делаю тестовые звонки. Раньше работало без проблем.

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "serg"<sip:1234@192.168.111.163>;tag=as7289b1f0

Звук появляется через минуту.

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 12:11
sm_sergey
А что здесь не так? Caleer id? Прописал, но ничего не изменилось. Звук появляется через минуту.

<--- SIP read from UDP:192.168.1.250:5060 --->
SIP/2.0 200 OK
From: "serg"<sip:1234@192.168.111.163>;tag=as288ad208
To: <sip:089516922533@192.168.1.250>;tag=776a028-fa01a8c0-13c4-55013-14c2fe7-10480a29-14c2fe7

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 12:22
Wapo
canreinvite=no ставили?

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 12:29
sm_sergey
Попробовал. То же самое. Звук спустя минуту.

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 13:18
ded
sm_sergey писал(а):А что здесь не так? Caleer id? Прописал, но ничего не изменилось. Звук появляется через минуту.
Вопрос не в Caleer id, а в адресации.
Если в заголовке SIP/2.0 200 OK
From: "Anonymous"<sip:anonymous@anonymous.invalid> то это разве правильно? Вот у меня будет эл. письмо не
From: "Сергей"<smazharov@gmail.com>
а From: "Anonymous"<anonymous@anonymous.invalid>, дойдёт по этому адресу какое-нить системное сообщение от MAILER-DAEMON о невозможности доставки почты, например?

Возможные причины такого поведения:
1) перегружен CPU
2) забился своп оперативной памяти
3) замедленно отрабатывает ДНС
4) локальные сети не прописаны в параметре localnet= и голос пытается сначала вязаться через роутер с помощью НАТ
5) .. (дополните сами)

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 13:47
Zavr2008
Reliably Transmitting (NAT) to 192.168.1.250:5060:
Итак NAT то совсем и не настроен.
- externip=
- localnet=
- nat=

Также в 1.8 уже нет canreinvite. Используем directmedia=no
exten => _08XXXXXXXXXX,1,Dial(SIP/192.168.1.250/${EXTEN},60,tTxX)
Не читали Будущее..
Пир называется trunk у Вас, так чего рисуем его IP вместо имени?

Также насчет анонимусов: попробуйте у транка sendrpid=yes, trustrpid=yes

Ну и напоследок: allowguest=yes - это для перчика чтоп все кому не лень имели в заднее отв? :)

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 14:10
sm_sergey
А что в настройках NAT прописывать?

Re: Звук при исходящем звонке появляется спустя некоторое вр

Добавлено: 29 янв 2015, 21:01
ded
Zavr2008 писал(а):Не читали Будущее..
Уже читать. Новичком на форуме считается прочитавший книгу "Будущее телефонии" и пытающийся сделать большее.
Часовой пояс: UTC+03:00
Страница 1 из 1

Создано на основе phpBB® Forum Software © phpBB Limited

Русская поддержка phpBB