PBX + cisco 2651xm + asterisk

glmonarch

Posted June 4, 2013 (edited)

Всем доброго дня.

У меня есть УПАТС Aastra MX-ONE TSW, подключенная к cisco 2651xm по E1 PRI. В свою очередь cisco 2651xm смотрит на asterisk по SIP-trunk. Я хочу использовать cisco 2651xm в качестве шлюза E1 PRI <==> SIP.

Звонки с софт-фона 4092, зарегистрированного на Asterisk, на номер 1312 (обычный телефон, подключенный к УПАТС) проходят нормально. А вот звонки в обратном направлении с 1312 на софт-фон 4092 не проходят - в трубке BUSY, в дебаге cisco2651xm следующие строки:

Jun 4 15:17:27.477: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

Jun 4 15:17:27.485: //299/B379E416804E/SIP/Event/sipSPICreateRpid: Received Octet3A=0x81 -> Setting ;screen=yes ;privacy=off

Jun 4 15:17:27.493: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:4092@10.2.210.13:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3BB

From: <sip:1312@172.26.18.200>;tag=1A74035E-E40

To: <sip:4092@10.2.210.13>

Date: Tue, 04 Jun 2013 15:17:27 GMT

Call-ID: B37D8ED0-CC6011E2-9F8C8019-37A47AC2@172.26.18.200

Supported: 100rel,timer,replaces

Min-SE: 1800

Cisco-Guid: 3011109910-3428848098-2152595473-2481374176

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: <sip:1312@172.26.18.200>;party=calling;screen=yes;privacy=off

Timestamp: 1370359047

Contact: <sip:1312@172.26.18.200:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 286

v=0

o=CiscoSystemsSIP-GW-UserAgent 1098 7100 IN IP4 172.26.18.200

s=SIP Call

c=IN IP4 172.26.18.200

t=0 0

m=audio 19354 RTP/AVP 8 0 101 19

c=IN IP4 172.26.18.200

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

Jun 4 15:17:27.497: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3BB;received=172.26.18.200;rport=53221

From: <sip:1312@172.26.18.200>;tag=1A74035E-E40

To: <sip:4092@10.2.210.13>;tag=as000066d4

Call-ID: B37D8ED0-CC6011E2-9F8C8019-37A47AC2@172.26.18.200

CSeq: 101 INVITE

Server: Asterisk PBX 11.2.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="000077af"

Content-Length: 0

Jun 4 15:17:27.501: //299/B379E416804E/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials

Jun 4 15:17:27.501: //299/B379E416804E/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge

Jun 4 15:17:27.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:4092@10.2.210.13:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3BB

From: <sip:1312@172.26.18.200>;tag=1A74035E-E40

To: <sip:4092@10.2.210.13>;tag=as000066d4

Date: Tue, 04 Jun 2013 15:17:27 GMT

Call-ID: B37D8ED0-CC6011E2-9F8C8019-37A47AC2@172.26.18.200

Max-Forwards: 70

CSeq: 101 ACK

Content-Length: 0

Router#

Jun 4 15:17:27.513: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

Cisco 2651xm имеет ip 172.26.18.200

Asterisk имеет ip 10.2.210.13

Меня смущает вот это сообщение из дебага:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3BB;received=172.26.18.200;rport=53221

From: <sip:1312@172.26.18.200>;tag=1A74035E-E40

To: <sip:4092@10.2.210.13>;tag=as000066d4

Правильно ли я понимаю, что cisco пытается зарегистрировать номер 1312 на asterisk? Если, да, то зачем? Как заставить ее корректно звонить на софт-фоны asterisk'а? Может у меня версия ios'а не подходящая для таких целей?

Router#sho ver

Cisco IOS Software, C2600 Software (C2600-ADVENTERPRISEK9-M), Version 12.4(25a), RELEASE SOFTWARE (fc2)

Technical Support: http://www.cisco.com/techsupport

Compiled Fri 22-May-09 21:06 by prod_rel_team

ROM: System Bootstrap, Version 12.2(8r) [cmong 8r], RELEASE SOFTWARE (fc1)

Router uptime is 5 days, 42 minutes

System returned to ROM by reload at 11:55:25 UTC Thu May 30 2013

System restarted at 12:00:38 UTC Thu May 30 2013

System image file is "flash:c2600-adventerprisek9-mz.124-25a.bin"

This product contains cryptographic features and is subject to United

States and local country laws governing import, export, transfer and

use. Delivery of Cisco cryptographic products does not imply

third-party authority to import, export, distribute or use encryption.

Importers, exporters, distributors and users are responsible for

compliance with U.S. and local country laws. By using this product you

agree to comply with applicable laws and regulations. If you are unable

to comply with U.S. and local laws, return this product immediately.

A summary of U.S. laws governing Cisco cryptographic products may be found at:

http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

If you require further assistance please contact us by sending email to

export@cisco.com.

Cisco 2651XM (MPC860P) processor (revision 3.1) with 253952K/8192K bytes of memory.

Processor board ID FOC08310NX3

M860 processor: part number 5, mask 2

2 FastEthernet interfaces

31 Serial interfaces

2 Channelized E1/PRI ports

32K bytes of NVRAM.

32768K bytes of processor board System flash (Read/Write)

Configuration register is 0x2102

Router#sho run

Building configuration...

Current configuration : 2126 bytes

! Last configuration change at 11:52:47 UTC Tue Jun 4 2013 by culture

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

hostname Router

boot-start-marker

boot-end-marker

card type e1 1 0

no logging console

no aaa new-model

no network-clock-participate slot 1

no network-clock-participate wic 0

ip cef

no ip domain lookup

ip auth-proxy max-nodata-conns 3

ip admission max-nodata-conns 3

isdn switch-type primary-qsig

voice-card 1

voice call carrier capacity active

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

redirect ip2ip

signaling forward rawmsg

sip

no call service stop

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

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glmonarch

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