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Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 09:58
graber
Здравствуйте!
Asterisk 1.6.0.3, centos 5.5, нахожусь за натом
Входящие звонки проходят нормально. При попытке совершить исходящий зовнок получаю Gateway Is Invalid.
Вот уже второй день бьюсь и не могу понять кто из нас инвалид, я или провайдер...
sip.conf
Код: Выделить всё
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
notifymimetype=text/plain
buggymwi=yes
users.conf
Код: Выделить всё
[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = yes
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
[discounttelecom]
secret=blablabla
username=74954444444
trunkname=discounttelecom
callerid=
hasexten=no
hassip=yes
hasiax=no
registeriax=no
registersip=yes
qualify=yes
host=sip.discounttelecom.ru
context=office
insecure=invite
fromuser=74954444444
fromdomain=sip.discounttelecom.ru
type=peer
contact=20001
disallow=all
allow=alaw
allow=ulaw
allow=g729
nat=yes
srvlookup=yes
externrefresh=10
canreinvite=no
dtmfmode=info
defaultexpiry=60
Вот что происходит при исходящем звонке(нахожусь не в офисе, поэтому звоню с 89032222222 на 84954444444 и набираю добавочный 701:
exten => 701,20,Dial(SIP/84991111111@discounttelecom,120,rTtWw)
Код: Выделить всё
-- Executing [701@office:10] Dial("SIP/74954444444-0896b730", "SIP/84991111111@discounttelecom,120,rTtWw") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.110.5.222 port 13844
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 81.94.129.162:5060:
INVITE sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK77655808;rport
Max-Forwards: 70
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as2e73acfa
To: <sip:84991111111@sip.discounttelecom.ru>
Contact: <sip:74954444444@10.110.5.222>
Call-ID: 1a2467950b3bc26f75a953ec504b8a63@sip.discounttelecom.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.3
Date: Thu, 02 Jun 2011 13:40:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 231
v=0
o=root 1007547094 1007547094 IN IP4 10.110.5.222
s=Asterisk PBX 1.6.0.3
c=IN IP4 10.110.5.222
t=0 0
m=audio 13844 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 84991111111@discounttelecom
-- Stopped music on hold on SIP/74954444444-0896b730
== Begin MixMonitor Recording SIP/74954444444-0896b730
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK77655808;rport=5060;received=213.108.1.11
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as2e73acfa
To: <sip:84991111111@sip.discounttelecom.ru>
Call-ID: 1a2467950b3bc26f75a953ec504b8a63@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 500 Gateway Is Invalid
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK77655808;rport=5060;received=213.108.1.11
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as2e73acfa
To: <sip:84991111111@sip.discounttelecom.ru>;tag=582417350-3759218061-201377213-1859863337
Call-ID: 1a2467950b3bc26f75a953ec504b8a63@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Reason: MVTSPRO;cause=40;text="Gateway Is Invalid"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 500 "Gateway Is Invalid" back from 81.94.129.162
Transmitting (NAT) to 81.94.129.162:5060:
ACK sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK77655808;rport
Max-Forwards: 70
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as2e73acfa
To: <sip:84991111111@sip.discounttelecom.ru>;tag=582417350-3759218061-201377213-1859863337
Contact: <sip:74954444444@10.110.5.222>
Call-ID: 1a2467950b3bc26f75a953ec504b8a63@sip.discounttelecom.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.3
Content-Length: 0
---
<< [ TYPE: Control (4) SUBCLASS: Congestion (8) ] [SIP/discounttelecom-0894b7b0]
-- SIP/discounttelecom-0894b7b0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [701@office:11] Hangup("SIP/74954444444-0896b730", "") in new stack
Ткните меня куда-нить носом, пожалуйста
Вот единственное что заметил, на другом сервере, с другим номером, но с тем же провайдером я в поле from вижу не 9032222222 а 89032222222
т.е. From: "89032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as2e73acfa
и звонок уходит...
Поставил x-lite, аккаунт рабочий, с исходящими всё в порядке
Получил ответ от провайдера:
Со своей стороны не видим проблемы для того что бы исходящие вызовы не ходили. Проблема в авторизации asterisk, sip-аккаунт и запрос invite не совпадают.
213.108.1.11.5060 > 81.94.129.162.5060: [udp sum ok] SIP, length: 523
OPTIONS sip:sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK483c3052;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.110.5.222>;tag=as4b1d20b9
To: <sip:sip.discounttelecom.ru>
Contact: <sip:asterisk@10.110.5.222>
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 10:51
ded
1) Перенесите провайдера
[discounttelecom]
secret=blablabla
username=74954444444
trunkname=discounttelecom
callerid=
hasexten=no
hassip=yes
hasiax=no
registeriax=no
registersip=yes
qualify=yes
host=sip.discounttelecom.ru
context=office
insecure=invite
fromuser=74954444444
fromdomain=sip.discounttelecom.ru
type=peer
contact=20001
disallow=all
allow=alaw
allow=ulaw
allow=g729
nat=yes ; это вряд ли...............
srvlookup=yes
externrefresh=10
canreinvite=no
dtmfmode=info
defaultexpiry=60
из users.conf в sip.conf и не надо ничего в users.conf. Эта штука для другой схемы предназначена.
и
2) Делайте наборы в формате
exten => 701,20,Dial(SIP/discounttelecom/84991111111,120,rTtWw)
а то в вашей конструкции да при юзере (!) discounttelecom (а это не юзер а пир!) ваш звонок совершается на SIP телефон, а не на SIP шлюз (gateway)
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 19:04
graber
Перенёс в sip.conf, убрал из users.conf
пробовал nat yes и no
ситуация та же...
sip.conf
Код: Выделить всё
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
notifymimetype=text/plain
buggymwi=yes
register => 74954444444:blablabla@sip.discounttelecom.ru/20001
[authentication]
[discounttelecom]
secret=blablabla
username=74954444444
trunkname=discounttelecom
callerid=
hasexten=no
hassip=yes
hasiax=no
registeriax=no
registersip=yes
qualify=yes
host=sip.discounttelecom.ru
context=office
insecure=invite
fromuser=74954444444
fromdomain=sip.discounttelecom.ru
type=peer
contact=20001
disallow=all
allow=alaw
allow=ulaw
allow=g729
nat=no
srvlookup=yes
externrefresh=10
canreinvite=no
dtmfmode=info
defaultexpiry=60
Код: Выделить всё
-- Executing [701@office:10] Dial("SIP/74954444444-094ef0d8", "SIP/discounttelecom/84991111111,120,rTtWw") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.110.5.222 port 17730
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 81.94.129.162:5060:
INVITE sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2e485a35;rport
Max-Forwards: 70
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as084cb861
To: <sip:84991111111@sip.discounttelecom.ru>
Contact: <sip:74954444444@10.110.5.222>
Call-ID: 001597a44d40b3177f2669c75206de4b@sip.discounttelecom.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.3
Date: Fri, 03 Jun 2011 14:58:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 192220690 192220690 IN IP4 10.110.5.222
s=Asterisk PBX 1.6.0.3
c=IN IP4 10.110.5.222
t=0 0
m=audio 17730 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called discounttelecom/84991111111
-- Stopped music on hold on SIP/74954444444-094ef0d8
== Begin MixMonitor Recording SIP/74954444444-094ef0d8
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2e485a35;rport=5060;received=213.108.1.11
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as084cb861
To: <sip:84991111111@sip.discounttelecom.ru>
Call-ID: 001597a44d40b3177f2669c75206de4b@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 500 Gateway Is Invalid
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2e485a35;rport=5060;received=213.108.1.11
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as084cb861
To: <sip:84991111111@sip.discounttelecom.ru>;tag=1860173773-3759272333-201362622-1859863337
Call-ID: 001597a44d40b3177f2669c75206de4b@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Reason: MVTSPRO;cause=40;text="Gateway Is Invalid"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 500 "Gateway Is Invalid" back from 81.94.129.162
Transmitting (no NAT) to 81.94.129.162:5060:
ACK sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2e485a35;rport
Max-Forwards: 70
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as084cb861
To: <sip:84991111111@sip.discounttelecom.ru>;tag=1860173773-3759272333-201362622-1859863337
Contact: <sip:74954444444@10.110.5.222>
Call-ID: 001597a44d40b3177f2669c75206de4b@sip.discounttelecom.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.3
Content-Length: 0
---
-- SIP/discounttelecom-09512790 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [701@office:11] Hangup("SIP/74954444444-094ef0d8", "") in new stack
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 19:25
ded
А ваш Астериск часом не за НАТом?
Если не указаны externip= то для корректного соединения будут проблемы.
Кроме того MERA MVTS3G v.4.2.0-27a может задробить такой звонок из-за его транзитности: CallerID его явно не должен быть таким - 9032222222.
From: "9032222222" <sip:74954444444@sip.discounttelecom.ru>;tag=as084cb861
To: <sip:84991111111@sip.discounttelecom.ru>;tag=1860173773-3759272333-201362622-1859863337
Это Вы там играетесь на звонках из транка снаружи - в другой транк - sip.discounttelecom.ru.
При обычном звонке с внутреннего номера через discounttelecom у Вас видать подставляется валидный CallerID, звонок проходит, а при транзитном - нет, и не проходит.
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 19:41
graber
Астериск за натом, externip указал..
Да, я в первом посте как раз и обращал внимание, что через другой номер этого же сип провайдера, такие звонки проходят, но в поле фром стоит 89032222222, а не 9032222222
В этом транке 4 линии, с 89032222222 я звоню на 8495444444, с этойго же номер уходит исходящий звонок.
при звонках с внутренних номеров, ситуация не меняется, т.е. вообще никакие исходящие не проходят.
А вот лог звонка с того же сервера, где я только что просто заменил username, secret и fromuser на другие, которые брал у дискаунттелекома пару месяцев назад:
Код: Выделить всё
-- Executing [701@office:10] Dial("SIP/74955555555-094eae60", "SIP/discounttelecom/84991111111,120,rTtWw") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.110.5.222 port 10292
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 81.94.129.162:5060:
INVITE sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK450044e3;rport
Max-Forwards: 70
From: "89032222222" <sip:74955555555@sip.discounttelecom.ru>;tag=as2cd11090
To: <sip:84991111111@sip.discounttelecom.ru>
Contact: <sip:74955555555@10.110.5.222>
Call-ID: 617b870b648586bf706358fd70460c61@sip.discounttelecom.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.3
Date: Fri, 03 Jun 2011 16:03:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 686182651 686182651 IN IP4 10.110.5.222
s=Asterisk PBX 1.6.0.3
c=IN IP4 10.110.5.222
t=0 0
m=audio 10292 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called discounttelecom/84991111111
-- Stopped music on hold on SIP/74955555555-094eae60
== Begin MixMonitor Recording SIP/74955555555-094eae60
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK450044e3;rport=5060;received=213.108.1.11
From: "89032222222" <sip:74955555555@sip.discounttelecom.ru>;tag=as2cd11090
To: <sip:84991111111@sip.discounttelecom.ru>
Call-ID: 617b870b648586bf706358fd70460c61@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
INFO sip:20001@10.110.5.222 SIP/2.0
Via: SIP/2.0/UDP 81.94.129.162:5060;rport;branch=z9hG4bK-2695843049-3759274637-201363646-18598633371
Via: SIP/2.0/UDP 81.94.129.162:5061;rport=5061;branch=z9hG4bK-2695843049-3759274637-201363646-1859863337;received=81.94.129.162
From: <sip:89032222222@81.94.129.162:5061;user=phone>;tag=1215612134-3759274637-201363646-1859863337
To: <sip:5555555@81.94.129.162;user=phone>;tag=as37e2dd20
Call-ID: e6c474668dfa11e0be90000c2943db6e@81.94.129.162
CSeq: 4 INFO
Contact: <sip:89032222222@81.94.129.162:5060>
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.2.0-27a
Content-Length: 26
Signal=1
Duration=270
<------------->
--- (12 headers 3 lines) ---
Receiving INFO!
<--- Transmitting (NAT) to 81.94.129.162:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.94.129.162:5060;branch=z9hG4bK-2695843049-3759274637-201363646-18598633371;received=81.94.129.162;rport=5060
Via: SIP/2.0/UDP 81.94.129.162:5061;rport=5061;branch=z9hG4bK-2695843049-3759274637-201363646-1859863337;received=81.94.129.162
From: <sip:89032222222@81.94.129.162:5061;user=phone>;tag=1215612134-3759274637-201363646-1859863337
To: <sip:5555555@81.94.129.162;user=phone>;tag=as37e2dd20
Call-ID: e6c474668dfa11e0be90000c2943db6e@81.94.129.162
CSeq: 4 INFO
User-Agent: Asterisk PBX 1.6.0.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:20001@10.110.5.222>
Content-Length: 0
<------------>
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK450044e3;rport=5060;received=213.108.1.11
From: "89032222222" <sip:74955555555@sip.discounttelecom.ru>;tag=as2cd11090
To: <sip:84991111111@sip.discounttelecom.ru>;tag=539185129-3759274637-201363646-1859863337
Call-ID: 617b870b648586bf706358fd70460c61@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Content-Type: application/sdp
Server: MERA MVTS3G v.4.2.0-27a
Content-Length: 240
v=0
o=- 1307116964 1307116964 IN IP4 81.94.129.162
s=-
c=IN IP4 81.94.129.162
t=0 0
m=audio 11632 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 81.94.129.162:11632
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 81.94.129.162:11632
-- SIP/discounttelecom-095009d0 is ringing
-- SIP/discounttelecom-095009d0 is making progress passing it to SIP/74955555555-094eae60
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK450044e3;rport=5060;received=213.108.1.11
From: "89032222222" <sip:74955555555@sip.discounttelecom.ru>;tag=as2cd11090
To: <sip:84991111111@sip.discounttelecom.ru>;tag=539185129-3759274637-201363646-1859863337
Call-ID: 617b870b648586bf706358fd70460c61@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Server: MERA MVTS3G v.4.2.0-27a
X-mera-expires: 3660
Content-Length: 240
v=0
o=- 1307116964 1307116964 IN IP4 81.94.129.162
s=-
c=IN IP4 81.94.129.162
t=0 0
m=audio 11632 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:84991111111@81.94.129.162:5060>
set_destination: Parsing <sip:84991111111@81.94.129.162:5060> for address/port to send to
set_destination: set destination to 81.94.129.162, port 5060
Transmitting (NAT) to 81.94.129.162:5060:
ACK sip:84991111111@81.94.129.162:5060 SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK40f51d5b;rport
Max-Forwards: 70
From: "89032222222" <sip:74955555555@sip.discounttelecom.ru>;tag=as2cd11090
To: <sip:84991111111@sip.discounttelecom.ru>;tag=539185129-3759274637-201363646-1859863337
Contact: <sip:74955555555@10.110.5.222>
Call-ID: 617b870b648586bf706358fd70460c61@sip.discounttelecom.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.3
Content-Length: 0
---
-- SIP/discounttelecom-095009d0 answered SIP/74955555555-094eae60
Всё проходит, всё хорошо. Вроде логично было бы предположить, что виноват провайдер, но почему же тогда работает софтфон? Да и в СТП мне говорят, что у них все настройки для всех номеров одинаковые.
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 20:27
ded
В пятницу вечером влом морщить извилины.
Чтобы не париться делаем так: превращаем
9032222222 в 89032222222
exten => 701,1,GotoIf(${CALLERID(num):0:1}=9?add8:allOK)
exten => 701,n(add8),Set(${CALLERID(num)}=8${CALLERID(num)})
exten => 701,n(allOK),Dial(SIP/discounttelecom/84991111111,120,rTtWw)
Может и напутал синтаксис, так не со зла. Думаю, мысль то ясна?
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 20:43
graber
Пофигу, всё так же:
Код: Выделить всё
-- Executing [701@office:6] Set("SIP/74955438968-b7e87950", "CALLERID(num)=89032222222") in new stack
-- Executing [701@office:7] Dial("SIP/74955438968-b7e87950", "SIP/discounttelecom/84991111111,120,rTtWw") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.110.5.222 port 16162
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 81.94.129.162:5060:
INVITE sip:84991111111@sip.discounttelecom.ru SIP/2.0
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2c4843c2;rport
Max-Forwards: 70
From: "89032222222" <sip:74955438968@sip.discounttelecom.ru>;tag=as51b575ab
To: <sip:84991111111@sip.discounttelecom.ru>
Contact: <sip:74955438968@10.110.5.222>
Call-ID: 60c9b39d4ea484f827678b0b15a3d420@sip.discounttelecom.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.3
Date: Fri, 03 Jun 2011 16:59:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 482086706 482086706 IN IP4 10.110.5.222
s=Asterisk PBX 1.6.0.3
c=IN IP4 10.110.5.222
t=0 0
m=audio 16162 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called discounttelecom/84991111111
-- Stopped music on hold on SIP/74955438968-b7e87950
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2c4843c2;rport=5060;received=213.108.1.11
From: "89032222222" <sip:74955438968@sip.discounttelecom.ru>;tag=as51b575ab
To: <sip:84991111111@sip.discounttelecom.ru>
Call-ID: 60c9b39d4ea484f827678b0b15a3d420@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
localhost*CLI>
<--- SIP read from UDP://81.94.129.162:5060 --->
SIP/2.0 500 Gateway Is Invalid
Via: SIP/2.0/UDP 10.110.5.222:5060;branch=z9hG4bK2c4843c2;rport=5060;received=213.108.1.11
From: "89032222222" <sip:74955438968@sip.discounttelecom.ru>;tag=as51b575ab
To: <sip:84991111111@sip.discounttelecom.ru>;tag=4201947586-3759211150-201364926-1859863337
Call-ID: 60c9b39d4ea484f827678b0b15a3d420@sip.discounttelecom.ru
CSeq: 102 INVITE
Contact: <sip:84991111111@81.94.129.162:5060>
Server: MERA MVTS3G v.4.2.0-27a
Reason: MVTSPRO;cause=40;text="Gateway Is Invalid"
Content-Length: 0
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 21:21
ded
В платный суппорт.
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 03 июн 2011, 21:29
graber
К дискаунттелекому? у них такого нет...
Если Вы возьмётесь - готов обсудить.
Re: Got SIP response 500 "Gateway Is Invalid"
Добавлено: 07 июн 2011, 08:35
graber
Спасибо, ded, проблема была со стороны провайдера, сейчас всё работает.