Help with setting Asterisk as Sip Client


#1

Have a user that wants to set his * server as a SIP client to my * server. Bonehead refuses to go with IAX, so I’m trying to test his configuration with my servers using docs from wiki. Codec on both ends is G723 (also tried g729a) codecs are allowed by each * server. Can get registerd fine. Problem when making calls is: WARNING[3799]: channel.c:2173 ast_channel_make_compatible: No path to translate from (local channel) to (remote channel). Any assistance appreciated. Thanks.


#2

Do you have a license or are you trying to use it as a pass thru?

info here

voip-info.org/tiki-index.php … isk+codecs


#3

pass thru. End clients using same codec (currently g723, but can change back to g729a). Client A dials Client B’s extension. Client B rings. Failure occurs on answer.

client A - g723 --> registered with serv A (allow g723, g729a) — SIP client to serv B
serv B (allow g723, g729a) --> client B - g723 (registerd with serv B)

sip.conf Serv A:
[general]
register => 997118867644:secret@<server_b ip address>/997118867644
port=5060
disallow=all
allow=g723
allow=g729a

[testclienta]
type=friend
username=testclienta
authuser=997118867644
fromuser=997118867644
secret=secret
callerid=<2905597>
accountcode=2905597
context=to-server-b
qualify=yes
host=dynamic
reinvite=yes
canreinvite=yes
nat=yes
tos=0x18

[server_b]
type=friend
secret=secret
username=997118867644
fromuser=997118867644
host=<server_b ip address>
nat=yes
context=from-server-b

extensions.conf serv A:
[to-server-b]
exten => _99X.,1,Dial(SIP/${EXTEN:2}@server_b)

Hope this gives a clearer picture. Thanks!


#4

noticed problem after last post… changed to allow g723.1 and allow g729 on serv A. Now getting one way voice path… any other pointers?


#5

Firewall?


#6

Sorry, I left that a little vague. I have partional success but I don’t think it is a firewall issue. Here are the call scenerios that work and the one I’m stuck on. I have serveral clients on *serv B that complete hundreds of calls through the cisco to the PSTN.

This works fine:
client A --> [firewall] --> *serv A --> *serv B --> [firewall] --> client B

So does this:
client B --> [firewall] --> *serv B --> *serv A --> [firewall] --> client A

and so does this:
client B --> [firewall] --> *serv B --> [firewall] --> cisco 3640 --> PSTN --> phone

But, this is one way voice (no voice back from phone to client A)
client A --> [firewall] --> *serv A --> *serv B --> [firewall] -->cisco 3640 --> PSTN --> phone

I’ll keep at it.

Thanks,
Russ


#7

Bingo… changed canreinvite to “no” and all seems well now. Thanks for you input, Rusty!

Thanks,
Russ


#8

glad you got it :laughing: