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Integration with asterisk solution

chingkiatlee
Level 1
Level 1

I plan to setup a normal integration with Asterisk (Free VoIP Solution) with Cisco Router 3725. The connection as below

CUCM -> (UTP) -> Cisco 3725 -> (E1 back to back cable) -> Asterisk Server -> (PRI) -> Asterisk Server outside

I do able to established the D-Channel connection with Asterisk Server locally. But when i do a call, the RTP failed to stream out.

The calling number can be display on the called user & vice-versa. But not the RTP package.

But what i found out is the Asterisk Server is only accepted G711.alaw codec (correct me if i am wrong) but my CUCM calling out is using G711.ulaw codec. So it there cause the problem?

CUCM version is 7.1 but the 3725 IOS SCCP only supported up to CUCM 3.4, so i failed to use the trancording.

So is anyone have the solution?

1 Accepted Solution

Accepted Solutions

acampbell
VIP Alumni
VIP Alumni

Hi,

Can you post the show run from your 3725 router.

Regards

Alex

Regards, Alex. Please rate useful posts.

View solution in original post

9 Replies 9

acampbell
VIP Alumni
VIP Alumni

Hi,

Can you post the show run from your 3725 router.

Regards

Alex

Regards, Alex. Please rate useful posts.

VoiceGW#sh run

Building configuration...

Current configuration : 2628 bytes

!

! Last configuration change at 12:45:51 MY Tue Nov 2 2010

! NVRAM config last updated at 12:45:55 MY Tue Nov 2 2010

!

version 12.3

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname HOST

!

boot-start-marker

boot-end-marker

!

enable secret 5 $1$okKg$J7Q0EIq8i1Jq3BmNsR5K1/

!

clock timezone MY 8

no network-clock-participate slot 1

network-clock-participate wic 0

network-clock-select 1 e1 0/0

voice-card 1

dspfarm

!

no aaa new-model

ip subnet-zero

ip cef

!

!

ip dhcp excluded-address 172.17.0.1 172.17.0.20

ip dhcp excluded-address 172.71.0.1 172.71.0.20

!

ip dhcp pool Voice

   network 172.17.0.0 255.255.248.0

   default-router 172.17.0.1

   option 150 ip 172.17.0.5

!

ip dhcp pool voice-new

   network 172.71.0.0 255.255.254.0

   default-router 172.71.0.1

   option 150 ip 172.71.0.4 172.71.0.5

!

no ftp-server write-enable

isdn switch-type primary-net5

!

!

!

voice service pots

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

!

!

!

!

!

!

!

!

!

!

!

controller E1 1/0

pri-group timeslots 1-31

description Connection to PSTN(MAXIS) 2888 E1 #1

!

controller E1 1/1

pri-group timeslots 1-31

description Connection to PSTN(MAXIS) 2828 E1 #2

!

!

!

interface FastEthernet0/0

ip address 172.17.0.8 255.255.248.0

duplex auto

speed auto

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Serial0/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-qsig

isdn timer T310 120000

isdn overlap-receiving

isdn protocol-emulate network

isdn incoming-voice voice

isdn send-alerting

isdn sending-complete

no cdp enable

!

interface Serial1/0:15

no ip address

isdn switch-type primary-net5

isdn incoming-voice voice

no cdp enable

!

interface Serial1/1:15

no ip address

isdn switch-type primary-net5

isdn incoming-voice voice

no cdp enable

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.17.0.1

ip http server

!

!

!

!

voice-port 0/0:15

echo-cancel coverage 32

timeouts interdigit 2

!

voice-port 1/0:15

echo-cancel coverage 32

cptone MY

timeouts initial 0

timeouts interdigit 2

!

voice-port 1/1:15

echo-cancel coverage 32

cptone MY

timeouts initial 0

timeouts interdigit 2

!

!

!

!

!

dial-peer voice 1000 pots

preference 10

destination-pattern 9T

progress_ind alert enable 8

direct-inward-dial

port 1/0:15

!

dial-peer voice 1001 pots

preference 1

destination-pattern 9T

progress_ind alert enable 8

direct-inward-dial

port 1/1:15

!

dial-peer voice 2000 voip

preference 10

destination-pattern .T

progress_ind setup enable 3

voice-class codec 1

session target ipv4:172.17.0.6

dtmf-relay h245-alphanumeric

!

dial-peer voice 2001 voip

preference 1

destination-pattern .T

progress_ind setup enable 3

voice-class codec 1

session target ipv4:172.17.0.5

dtmf-relay h245-alphanumeric

!

dial-peer voice 3000 pots

preference 6

destination-pattern 0T

progress_ind alert enable 8

direct-inward-dial

port 0/0:15

!

!

line con 0

line aux 0

line vty 0 4

password cisco123

login

!

ntp clock-period 17180644

ntp master

ntp server 172.16.10.21

end

Hi acampbell,

FYI.

The D-channel is working fine & the signalling do pass thru, only the RTP does not work.

Not audio for both side.

ashok_boin
Level 5
Level 5

Hi,

Asterisk do support other codecs.

http://www.voip-info.org/wiki/view/Asterisk+codecs

Regards...

-Ashok.


With best regards...
Ashok

Hi ashok,

Do this supported for Qsig connection?

Cause based on the engineer, he do said it only supported for SIP trunk.

Correct me if i am wrong.

Hi Lee,

Yes, Asterisk does support QSIG a long back.

Pls see the article in 2005 published in network world magazine.

http://www.networkworld.com/news/2005/111605-asterisk-voip.html

Can you please post any debugs from 3725 to Asterisk (signaling + media)? And also, what's the relevant config in the above shared config for Asterisk server including interface?

Regards...

-Ashok.


With best regards...
Ashok

Mar 22 03:24:51.996: //-1/801AAC6A2C00/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=2027

   ----- ccCallInfo IE subfields -----

   cisco-ani=2027

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=1234

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=-1

   cisco-rdnplan=-1

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1

Mar 22 03:24:51.996: //-1/801AAC6A2C00/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x6509D0E4, Call Info(

   Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=1234(TON=Unknown, NPI=Unknown),

   Calling Translated=FALSE, Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=2002, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=7

Mar 22 03:24:51.996: //-1/801AAC6A2C00/CCAPI/ccCheckClipClir:

   In: Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

Mar 22 03:24:51.996: //-1/801AAC6A2C00/CCAPI/ccCheckClipClir:

   Out: Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

Mar 22 03:24:51.996: //7/801AAC6A2C00/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=1234(TON=Unknown, NPI=Unknown))

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/cc_process_call_setup_ind:

   Event=0x6530F668

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCallSetContext:

   Context=0x658BDE60

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 7 with tag 2002 to app "_ManagedAppProcess_Default"

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCallSetupRequest:

   Destination=, Calling IE Present=TRUE, Mode=0,

   Outgoing Dial-peer=3000, Params=0x658BEAF0, Progress Indication=NULL(0)

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCheckClipClir:

   In: Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCheckClipClir:

   Out: Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCallSetupRequest:

   Destination Pattern=1234, Called Number=1234, Digit Strip=TRUE

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/ccCallSetupRequest:

   Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=1234(TON=Unknown, NPI=Unknown),

   Redirect Number=, Display Info=

   Account Number=2027, Final Destination Flag=TRUE,

   Guid=801AAC6A-92E2-D1E9-2C00-B903AC1901BC, Outgoing Dial-peer=3000

Mar 22 03:24:52.000: //7/801AAC6A2C00/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=2027

   ----- ccCallInfo IE subfields -----

   cisco-ani=2027

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=1234

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=-1

   cisco-rdnplan=-1

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1

Mar 22 03:24:52.004: //7/801AAC6A2C00/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x6534EB18, Interface Type=6, Destination=, Mode=0x0,

   Call Params(Calling Number=2027(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=1234(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=3000, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

Mar 22 03:24:52.004: //8/801AAC6A2C00/CCAPI/ccIFCallSetupRequestPrivate:

   SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1

Mar 22 03:24:52.004: //8/801AAC6A2C00/CCAPI/ccCallSetContext:

   Context=0x658BEAA0

Mar 22 03:24:52.004: //7/801AAC6A2C00/CCAPI/ccSaveDialpeerTag:

   Outgoing Dial-peer=3000

Mar 22 03:24:52.004: ISDN Se0/0:15 Q931: Applying typeplan for sw-type 0x16 is 0x0 0x0, Calling num 2027

Mar 22 03:24:52.008: ISDN Se0/0:15 Q931: Applying typeplan for sw-type 0x16 is 0x0 0x0, Called num 1234

Mar 22 03:24:52.008: ISDN Se0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0083

        Sending Complete

This is not helping much.

Can you pls try with "debug voip ccapi inout" and share the output? If needed, you can perform "debug vtsp rtp" command to debug RTP packet debugging later.

Regards...

-Ashok.


With best regards...
Ashok

Hi Ashok,

This is the debug voip ccapi inout result right?

below is the debug isdn q931 result.

Mar 22 03:32:49.601: ISDN Se1/1:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Calling num 2027

Mar 22 03:32:49.601: ISDN Se1/1:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Called num <>

Mar 22 03:32:49.601: ISDN Se1/1:15 Q931: TX -> SETUP pd = 8  callref = 0x008A

        Bearer Capability i = 0x8890

                Standard = CCITT

                Transfer Capability = Unrestricted Digital

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA9838F

                Exclusive, Channel 15

        Calling Party Number i = 0x0081, '2027'

                Plan:Unknown, Type:Unknown

        Called Party Number i = 0x80, '<>'

                Plan:Unknown, Type:Unknown

Mar 22 03:32:49.637: ISDN Se1/1:15 Q931: RX <- SETUP_ACK pd = 8  callref = 0x808A

        Channel ID i = 0xA9838F

                Exclusive, Channel 15

Mar 22 03:32:49.813: ISDN Se1/1:15 Q931: RX <- CALL_PROC pd = 8  callref = 0x808A

Mar 22 03:32:50.865: ISDN Se1/1:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x808A

        Cause i = 0x8090 - Normal call clearing

Mar 22 03:32:50.869: ISDN Se1/1:15 Q931: TX -> RELEASE pd = 8  callref = 0x008A

Mar 22 03:32:50.897: ISDN Se1/1:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x808A