Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас
Задайте Ваш вопрос

Snom meetingpoint исходящие звонки

0

Всем старожилам привет :) Купил в конф зал вот это Snome Meetingpoint установил и завел новую учетку на уже отлично работающем Астериске. Ситуация такая что телефон регается на * и и принимает вхоядящие звонки а вот исходящие не работают...(матюки)... в веб морде телефона есть трасировочка и это очень удобно. Удаляю всю трассировку и делаю звонок на внутренний номер 100(* 1.1, телефон 1.238):

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:18:32:739 (1116 bytes):
INVITE sip:100@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-bjg1mtxlvga8;rport
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
X-Serialnumber: 000413322606
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snomMP/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 688959849 688959849 IN IP4 192.168.1.238
s=call
c=IN IP4 192.168.1.238
t=0 0
m=audio 53216 RTP/AVP 0 8 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hqkVo7IUww4B2imn3gd9LNtBfqVt9EO298nO9vSg
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=ptime:20
a=sendrecv
Received from udp:192.168.1.1:5060 at 15/9/2014 15:18:32:748 (523 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-bjg1mtxlvga8;received=192.168.1.238;rport=45159
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>;tag=as109161f3
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7413f7c3"
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:18:32:755 (373 bytes):

ACK sip:100@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-bjg1mtxlvga8;rport
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>;tag=as109161f3
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:18:32:768 (1283 bytes):

INVITE sip:100@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-q2crdyfpseez;rport
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
X-Serialnumber: 000413322606
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snomMP/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username="110",realm="asterisk",nonce="7413f7c3",uri="sip:100@192.168.1.1;user=phone",response="1acdb1d7675f412e01a930e041a6ed02",algorithm=MD5
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 688959849 688959849 IN IP4 192.168.1.238
s=call
c=IN IP4 192.168.1.238
t=0 0
m=audio 53216 RTP/AVP 0 8 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hqkVo7IUww4B2imn3gd9LNtBfqVt9EO298nO9vSg
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=ptime:20
a=sendrecv
Received from udp:192.168.1.1:5060 at 15/9/2014 15:18:32:858 (454 bytes):

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-q2crdyfpseez;received=192.168.1.238;rport=45159
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>;tag=as109161f3
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:18:32:864 (373 bytes):

ACK sip:100@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.238:45159;branch=z9hG4bK-q2crdyfpseez;rport
From: "RR PBX" <sip:110@192.168.1.1>;tag=261jyjj5a8
To: <sip:100@192.168.1.1;user=phone>;tag=as109161f3
Call-ID: 18d91654a4a6-1cjbchvqyk6y
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Content-Length: 0

вижу что ломится на 5060 по протоколу udp О.о с какого? в настройках так и не нашел где можно указать что сигнализация работает через 5060 TCP. Можно конечно забиндить попробовать на астериске SIP и на udp я думаю но хотелось бы попробовать решить проблему без этого. Сталкивался кто с таким телефоном?

sip трассировка входящего звонка который нормально проходит и тут так же сип по udp и ничего:

Received from udp:192.168.1.1:5060 at 15/9/2014 15:32:26:028 (881 bytes):

INVITE sip:110@192.168.1.238:45159;line=hd1s1t15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport
Max-Forwards: 70
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>
Contact: <sip:107@192.168.1.1:5060>
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Mon, 15 Sep 2014 12:34:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 1277113151 1277113151 IN IP4 192.168.1.1
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.1.1
t=0 0
m=audio 18056 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:26:046 (374 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:26:071 (517 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:26:573 (517 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:27:573 (517 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Received from udp:192.168.1.1:5060 at 15/9/2014 15:32:28:647 (381 bytes):

CANCEL sip:110@192.168.1.238:45159;line=hd1s1t15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport
Max-Forwards: 70
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:28:651 (307 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 CANCEL
Content-Length: 0

Sent to udp:192.168.1.1:5060 at 15/9/2014 15:32:28:656 (386 bytes):

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport=5060
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 INVITE
Contact: <sip:110@192.168.1.238:45159;line=hd1s1t15>;reg-id=1
Content-Length: 0

Received from udp:192.168.1.1:5060 at 15/9/2014 15:32:28:736 (427 bytes):

ACK sip:110@192.168.1.238:45159;line=hd1s1t15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK342a1337;rport
Max-Forwards: 70
From: "Bu" <sip:107@192.168.1.1>;tag=as3a31f795
To: <sip:110@192.168.1.238:45159;line=hd1s1t15>;tag=gr7e1z1iil
Contact: <sip:107@192.168.1.1:5060>
Call-ID: 0a06da4942bcc78a497813e803f53738@192.168.1.1:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0
удалить закрыть спам изменить тег редактировать

спросил 2014-09-15 16:24:10 +0400

DJs3000 Gravatar DJs3000 flag of Russian Federation
551 41 13 33
http://retrogamesbattle.c...

обновил 2014-09-15 16:40:41 +0400

Comments

Может кому пригодится :) вообщем после пары дней бадания с телефоном я таки нашел в чем проблема была. В настройке учетки стаяло криптование :) убираем криптование sip канала в настройках конкретной учети и все нормуль.

DJs3000 ( 2014-12-18 04:06:38 +0400 )редактировать

1 Ответ

1

By default the phone isn't listening on port 5060 for TCP connections, to change this

behaviour, enable this option.

ATTENTION: This option will not be available any more in 8.7.4 versions and higher!

XML CONFIGURATION

< tcplisten perm="PERMISSIONFLAG" > VALIDVALUE < /tcplisten >

VALIDVALUE

< on >, < off >

DEFAULTVALUE

off

Оно? ;-)

ссылка удалить спам редактировать

ответил 2014-09-16 09:29:10 +0400

bolshoy_plohish Gravatar bolshoy_plohish
1358 21 16 37

Comments

Так-то оно но в веб морде нет такой опции :) видимо нужно конфиг подгружать через какой TFTP нада будет попробовать

DJs3000 ( 2014-09-17 14:28:26 +0400 )редактировать

Ваш ответ

Please start posting your answer anonymously - your answer will be saved within the current session and published after you log in or create a new account. Please try to give a substantial answer, for discussions, please use comments and please do remember to vote (after you log in)!
[скрыть предварительный просмотр]

Закладки и информация

Добавить закладку

подписаться на rss ленту новостей

Статистика

Задан: 2014-09-15 16:24:10 +0400

Просмотрен: 367 раз

Обновлен: Sep 16 '14

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