asked on

Asterisk Voice Connectivity with Cisco AS5350 via H323

Hello everyone,

I am little confused with the following scenario

I have created Dial peer on AS5350 which is use to take call from pstn and send to asterisk PBX using H323 for incoming call with ulaw and alaw voice codecs , I have installed asterisk-now downloaded from asterisk.org , I am getting successfully call hit on asterisk from cisco , but when i configure asterisk to transfer call to specific extension than there is no voice on call after signal hit, means asterisk is receiving signals but not media from cisco or might be something wrong in h323.conf file , I have tested with same call hit with nortel PBX , there is no single issue in it

so what i did on asterisk in extensions.conf

note :- 123456 is suppose to be my incoming number from cisco access server

[default]
exten => 123456,1,Dial(SIP/100,25,tT)

once I saved this configuration using reload on asterisk -r , I start getting calls on 100 extension but no voice connectivity

now what kind of input you want from my side to solve this problem ASAP

Cheers:)
Aamir

icenick

Dear aamir_inayat,

Since the call does hit your asterisk server, can you post here that portion of asterisk terminal?

Moreover, can you post your sip.conf configuration?

I suggest also you should download and install Wireshark. It's a packet sniffing tool that will help you capture calls and detect errors and faults at the protocol level.

icenick

You can find wireshark here:

http://www.wireshark.org/

aamir_inayat

ASKER

Dear ICENICK,

please find attached files

in this file you can find configuration for sip.conf ,extension.conf , h323.conf ,sip_notify.conf

as far call concern , it is hitting on asterisk , as you can find below

---------------------
-- Executing [123456@default:1] Goto("H323/ip$10.100.250.90:50468/18613", "voicemenu-custom-1|s|1") in new stack
-- Goto (voicemenu-custom-1,s,1)
-- Executing [s@voicemenu-custom-1:1] Answer("H323/ip$10.100.250.90:50468/18613", "") in new stack
-- Executing [s@voicemenu-custom-1:2] Wait("H323/ip$10.100.250.90:50468/18613", "1") in new stack
-- Executing [s@voicemenu-custom-1:3] BackGround("H323/ip$10.100.250.90:50468/18613", "thank-you-for-calling") in new stack
-- <H323/ip$10.100.250.90:50468/18613> Playing 'thank-you-for-calling' (language 'en')
-- Executing [s@voicemenu-custom-1:4] BackGround("H323/ip$10.100.250.90:50468/18613", "if-u-know-ext-dial") in new stack
-- <H323/ip$10.100.250.90:50468/18613> Playing 'if-u-know-ext-dial' (language 'en')
-- Executing [s@voicemenu-custom-1:5] BackGround("H323/ip$10.100.250.90:50468/18613", "otherwise") in new stack
-- <H323/ip$10.100.250.90:50468/18613> Playing 'otherwise' (language 'en')
-- Executing [s@voicemenu-custom-1:6] BackGround("H323/ip$10.100.250.90:50468/18613", "to-reach-operator") in new stack
-- <H323/ip$10.100.250.90:50468/18613> Playing 'to-reach-operator' (language 'en')
-- Executing [s@voicemenu-custom-1:7] BackGround("H323/ip$10.100.250.90:50468/18613", "pls-hold-while-try") in new stack
-- <H323/ip$10.100.250.90:50468/18613> Playing 'pls-hold-while-try' (language 'en')
[Dec 17 14:28:32] WARNING[3969]: pbx.c:2514 __ast_pbx_run: Invalid extension '501', but no rule 'i' in context 'voicemenu-custom-1'

----------------------

here on asterisk console it shows that call is coming but there is not voice in call

for packet traces , I am going to attach the following trace for using tcpdump

tcpdump -i eth0 -vns0 not port 22 -w file.cap -- if you want anyother arguments let me know

please rename the file file.xls to file.pcap to open in to wireshark

if you need more input let me know

:D
Config.docx
file.xls

icenick

Dear aamir_inayat,

From asterisk console, I can see that it plays the sounds files.

To get this right, you can not hear the sounds playing? Right?

You have to configure the cisco gateway as a sip/h.323 user in sip.conf/h323.conf. Where is the setup. I could not find the related part in the config file (sip.conf, etc.).

Here is what I would do:

sip.conf:

[cisco]
host=cisco_IP_ADDRESS
type=friend
context=default
insecure=very ; This is important, you should put it

extensions.conf:

[default]

...

exten => 123456,1,Background(thank-you-for-calling)

...

I suggest before running the whole dial plan is to go simple first. Try the above dial plan, just to play a simple message.

Good luck.

aamir_inayat

ASKER

let me check it and update you
one more thing that my asterisk machine is behind nat , so how to enable natting for h323 on asterisk

icenick

I am not familiar with h323 nat issues on asterisk.

Read those regarding sip and I am sure they will give a bit of what should be done:

http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html

http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

Good luck.

aamir_inayat

ASKER

tried but still not working , what else it could be

aamir_inayat

ASKER

Guys ,

I have removed system from nat , now Access server and asterisk are on same network , but still no voice connectivity as i explained above , DTMF is working but no Voice connectivity between two end points from AS to asterisk

looks like its problem with asterisk

Bilal

icenick

Dear aamir_inayat,

Can't think of anything at the moment. Have you tried disabling the firewall?

aamir_inayat

ASKER

Dear Icenick,

yes i did it. please find below my senior below as i am working now.

{PSTN} ----ss7 Link---{My office ASA5300}---------Ethernet Link------- {Asterisk}------Ethernet link----IP Phone. so there is not firewall.

i want only incomming call for this server. call is coming but no connection voice when i accept call from my mobile or other pstn .

I have nortel PBX also. when i connect Nortel PBX instead of Asterisk . everything is working.

Thanks for your help..

ASKER CERTIFIED SOLUTION

icenick

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