cisco as5300 --> asterisk

jlorozco1975 · ‎09-10-2012

Hi

Good day everyone

I need your help please.

i need to connect my cisco as5300 to a third party cel provider with asterisk.

something like this

my agent dial a cel phone number-->cisco as5300-->internet-->asterisk-->final user (more less)

here is my configuration of the dial-peers:

!

dial-peer voice 200 voip

description LINK TEST AST

huntstop

destination-pattern .

voice-class codec 1

session protocol sipv2

session target ipv4:187.162.129.75:5060

!

dial-peer voice 201 pots

description LINK TEST AST

preference 1

max-conn 30

destination-pattern .

direct-inward-dial

port 1:1

!

i have an ACL

access-list 1 permit 187.162.129.75

And this is what it shows me with show call history voice brief

Total call-legs: 2

1968 : 9343518hs.92 +-1 +31 pid:200 Originate 0444423338885

dur 00:00:00 tx:0/0 rx:0/0 39 (bearer capability not authorized)

IP 187.162.129.75:0 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8

1968 : 9343518hs.93 +-1 +105 pid:201 Answer . <-----(it supose that here must be the dialed number)

dur 00:00:00 tx:0/0 rx:0/0 39 (bearer capability not authorized)

Telephony 1:1:92: tx:0/0/0ms None noise:0dBm acom:0dBm

On the asterisk side, what the cisco sends is only a "."

I can't find where i am wrong.

Please help me

Thanks in advance

Kind regards.

Chris Deren · ‎09-10-2012

Your destination pattern collects only single sigit, if you want it to be a wildcard to collect any digit then change the following:

dial-peer voice 200 voip

destination-pattern .T

Since you did not post full config we cannot see if there are other issues, here so if that does not fully resolve the issue provide "sh run", "sh ver" and "debug ccsip messages" for a call.

HTH,

Chris

jlorozco1975 · ‎09-10-2012

Sorry

Here is the config

Building configuration...

Current configuration : 6068 bytes

!

! Last configuration change at 17:34:09 CST Mon Sep 10 2012

! NVRAM config last updated at 15:58:50 CST Mon Sep 10 2012

!

version 12.2

no parser cache

no service single-slot-reload-enable

service timestamps debug datetime msec localtime

service timestamps log datetime msec localtime

no service password-encryption

!

hostname GWVOIP_IMPULSE3

!

logging rate-limit console 10 except errors

enable secret

enable password

!

resource-pool disable

clock timezone CST -6

clock summer-time CST recurring

clock calendar-valid

!

ip subnet-zero

no ip domain-lookup

!

no ip dhcp-client network-discovery

voice call send-alert

voice rtp send-recv

!

voice class codec 1

codec preference 1 g729r8 bytes 80

codec preference 2 g729br8 bytes 80

codec preference 3 g723r63 bytes 48

codec preference 4 g723r53 bytes 40

codec preference 5 g723ar63 bytes 48

codec preference 6 g723ar53 bytes 40

!

voice class h323 1

h225 timeout tcp establish 10

!

fax interface-type vfc

mta receive maximum-recipients 0

!

controller E1 0

framing NO-CRC4

clock source line secondary 1

ds0-group 1 timeslots 1-7 type r2-digital r2-compelled

ds0-group 2 timeslots 8-12 type r2-digital r2-compelled

ds0-group 3 timeslots 22-28 type r2-digital r2-compelled

ds0-group 4 timeslots 30-31 type r2-digital r2-compelled

ds0-group 5 timeslots 20-21 type r2-digital r2-compelled

ds0-group 6 timeslots 13-15,17-19 type r2-digital r2-compelled

ds0-group 7 timeslots 29 type r2-digital r2-compelled

cas-custom 1

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 2

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 3

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 4

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 5

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 6

country telmex use-defaults

category 2

answer-signal group-b 1

cas-custom 7

country telmex use-defaults

category 2

answer-signal group-b 1

!

controller E1 1

framing NO-CRC4

ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled

cas-custom 1

country telmex use-defaults

category 2

answer-signal group-b 1

!

controller E1 2

framing NO-CRC4

ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled

cas-custom 1

country telmex use-defaults

category 2

answer-signal group-b 1

!

controller E1 3

framing NO-CRC4

clock source line primary

ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled

cas-custom 1

country telmex use-defaults

category 2

answer-signal group-b 1

!

gw-accounting h323

gw-accounting h323 vsa

gw-accounting voip

!

interface Ethernet0

no ip address

shutdown

!

interface FastEthernet0

ip address

duplex auto

speed auto

!

ip default-gateway

ip classless

ip route 0.0.0.0 0.0.0.0

no ip http server

ip rtcp report interval 10000

!

logging history warnings

access-list 3 permit 189.180.47.135

!

tftp-server system:

!

call rsvp-sync

!

voice-port 0:1

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

playout-delay mode adaptive

cptone MX

bearer-cap Speech

!

voice-port 0:2

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

playout-delay mode adaptive

cptone MX

bearer-cap Speech

!

voice-port 0:3

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

playout-delay mode adaptive

cptone MX

bearer-cap Speech

!

voice-port 0:4

compand-type a-law

!

voice-port 0:5

compand-type a-law

!

voice-port 0:6

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

playout-delay mode adaptive

cptone MX

bearer-cap Speech

!

voice-port 0:7

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

playout-delay mode adaptive

cptone MX

bearer-cap Speech

!

voice-port 1:1

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

cptone MX

bearer-cap Speech

!

voice-port 2:1

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

cptone MX

bearer-cap Speech

!

voice-port 3:1

input gain 2

output attenuation 2

echo-cancel coverage 24

compand-type a-law

cptone MX

bearer-cap Speech

!

mgcp profile default

!

dial-peer voice 200 voip

description LINK TEST AST

huntstop

destination-pattern .T

voice-class codec 1

session protocol sipv2

session target ipv4:187.162.129.75:5060

!

dial-peer voice 201 pots

description LINK TEST AST

preference 1

max-conn 30

destination-pattern .

direct-inward-dial

port 1:1

!

line con 0

password

line aux 0

line vty 0 4

access-class 3 in

exec-timeout 30 0

password

logging synchronous

login

!

ntp clock-period 17180058

ntp update-calendar

ntp server 132.163.4.102

ntp server 131.107.1.10

ntp server 129.6.15.28

end

Here is the version

Cisco Internetwork Operating System Software

IOS (tm) 5300 Software (C5300-IS-M), Version 12.2(xa3plus.0.10.0), CISCO DEVELOPMENT TEST VERSION

Compiled Tue 19-Mar-02 13:10 by hwcheng

Image text-base: 0x60008950, data-base: 0x60FF8000

ROM: System Bootstrap, Version 12.0(2)XD1, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1)

BOOTFLASH: 5300 Software (C5300-BOOT-M), Version 12.2(2)XA3, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1)

GWVOIP_IMPULSE3 uptime is 1 day, 3 hours, 36 minutes

System returned to ROM by reload at 13:34:31 CST Wed Jul 20 2011

System restarted at 14:21:45 CST Sun Sep 9 2012

System image file is "flash:c5300-is-mz-xa3plus.0.10.0"

cisco AS5300 (R4K) processor (revision A.32) with 131072K/16384K bytes of memory.

Processor board ID 13228260

R4700 CPU at 150Mhz, Implementation 33, Rev 1.0, 512KB L2 Cache

Channelized E1, Version 1.0.

Bridging software.

X.25 software, Version 3.0.0.

Primary Rate ISDN software, Version 1.1.

Backplane revision 2

Manufacture Cookie Info:

EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x30,

Board Hardware Version 1.80, Item Number 800-2544-03,

Board Revision A0, Serial Number 13228260,

PLD/ISP Version 0.0, Manufacture Date 20-Apr-1999.

1 Ethernet/IEEE 802.3 interface(s)

1 FastEthernet/IEEE 802.3 interface(s)

4 Channelized E1/PRI port(s)

60 Voice resource(s)

128K bytes of non-volatile configuration memory.

16384K bytes of processor board System flash (Read/Write)

8192K bytes of processor board Boot flash (Read/Write)

Configuration register is 0x2102

And the debug

GWVOIP_IMPULSE3#

Sep 10 18:01:39.900: Sent:

INVITE sip:0444423338885@187.162.129.75:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060

From: "." <.>xxx.xxx.xxx.xxx>;tag=5EFB2E4-FC4

To: <0444423338885>

Date: Mon, 10 Sep 2012 18:01:39 GMT

Call-ID: 50879E25-FAD211E1-880DC957-CCDECCA9@200.76.37.179

Cisco-Guid: 1351025149-4208071137-2282473815-3437153449

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

CSeq: 101 INVITE

Max-Forwards: 6

Timestamp: 1347318099

Contact: <.>xxx.xxx.xxx.xxx:5060;user=phone>

Expires: 180

Content-Type: application/sdp

Content-Length: 138

v=0

o=CiscoSystemsSIP-GW-UserAgent 898 5676 IN IP4 xxx.xxx.xxx.xxx

s=SIP Call

c=IN IP4 xxx.xxx.xxx.xxx

t=0 0

m=audio 18496 RTP/AVP 18 4

Sep 10 18:01:39.976: Received:

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;

received=xxx.xxx.xxx.xxx;rport=53320

From: "." <.>;tag=5EFB2E4-FC4

To: <0444423338885>;tag=as16c91089

Call-ID: 50879E25-FAD211E1-880DC957-CCDECCA9@xxx.xxx.xxx.xxx

CSeq: 101 INVITE

Server: FPBX-2.8.1(1.8.11.0)

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="592df766"

Content-Length: 0

GWVOIP_IMPULSE3#

Sep 10 18:01:39.980: Sent:

ACK sip:0444423338885@187.162.129.75:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060

From: "." <.>;tag=5EFB2E4-FC4

To: <0444423338885>;tag=as16c91089

Date: Mon, 10 Sep 2012 18:01:39 GMT

Call-ID: 50879E25-FAD211E1-880DC957-CCDECCA9@xxx.xxx.xxx.xxx

Max-Forwards: 6

Content-Length: 0

CSeq: 101 ACK

Chris Deren · ‎09-10-2012

The clue here is

SIP/2.0 401 Unauthorized

Looks like your provider requires authentication to be enabled for the SIP trunk but I do not see any sip-ua configuration with authentication in your config.

Did they provide you with SIP trunk autntication username and password?

HTH,

Chris

jlorozco1975 · ‎09-10-2012

No, they just send me the ip add, the codec and thats all.

i'm gonna check with them

Thanks in advance.

Regards.

Chris Deren · ‎09-11-2012

Great, let us know what you find out.

HTH, please rate all useful posts!

Chris

jlorozco1975 · ‎09-13-2012

Good day

Just for the record, it's working now, the problem was on the asterisk side

GWVOIP_IMPULSE3#sh voi call sum

PORT CODEC VAD VTSP STATE VPM STATE

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

1:1.28 g729r8 y S_CONNECT CSM_IC7_CONNECTED

Thanks a lot.

Regards.

stephan_maurer · ‎02-22-2013

Hi Juan,

great info.

could you please describe what hardware are you using on the AS5300? E1 Card/ VoIP Card, DSPs?

Thanks a lot.

Regards

hfb5 · ‎10-09-2012

Hola Juan, esto es por el codec, el voice class codec que estas colocando no esta siendo aceptado por el asterisk, solo por probar en el dial-peer voip quita el voice class codec y coloca codec g711allow.

y llamas de nuevo!

Cordialmente

Heiber