Asterisk with Proxy registrer problem

Asterisk Asterisk Support
1

Hi there,

i have an SIP-Proxy (Innovaphone IP302). IP 172.16.5.51
there is a phone configured with the following parameter
user = 702
secret = 0

if i dial from a softphone direct on this proxy e.g the nr 626 everything works fine

now i try to use the asterisk instead of the softphone

i connect the phone to the asterisk as SIP/2000

[2000]
type=friend
context=phone
secret=2000
host=dynamic

that works fine

and then i try to dial from the phone 626 to do this via the proxy

but i don’t know how?

i try some examples from some asterisk book and search the web but nothing works.

currently mit extensions.conf looks like:

[phone] exten => 702,1,Dial(SIP/702) exten => 626,1,Dial(SIP/626)

and my sip.conf

[2000]
type=friend
context=phone
secret=2000
host=dynamic

[626]
type=peer
context=phone
host=172.16.5.51
fromuser=702
username=702
secret=0

and the error i get is

i guess i did something terribly wrong… but i don’t have any further ideas

i also try to find a solution in http://www.the-asterisk-book.com/unstable/ but without success…

maybe someone can help me pls :smile:

Oli

2

would need to see a full SIP trace to see what is happening. issue a “sip set debug” at the asterisk CLI and post the results here.

3

hm i have only an Asterisk 1.2 :frowning: …

but here is the output on screen

[quote] – Executing Dial(“SIP/2000-081a0b90”, “SIP/626”) in new stack
– Called 626
Aug 14 14:44:23 WARNING[4634]: chan_sip.c:9923 handle_response_invite: Forbidden - wrong password on authentication for INVITE to ‘“unknown” sip:702@172.16.5.209;tag=as211f3700’
– SIP/626-081a6220 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Aug 14 14:44:33 WARNING[5093]: pbx.c:2416 __ast_pbx_run: Timeout, but no rule ‘t’ in context ‘phone’[/quote]

if this will help you ?

otherwise i need to install 1.4 or 1.6

4

installing 1.4 may fix your problem anyway :smile: You could always use tcpdump to capture all the traffic on UDP 5060 to get the equivalent of a sip trace.

without seeing what’s going on under the hood, i can’t really help you fix the issue.

5

[quote=“g2010”]installing 1.4 may fix your problem anyway Smile You could always use tcpdump to capture all the traffic on UDP 5060 to get the equivalent of a sip trace.

without seeing what’s going on under the hood, i can’t really help you fix the issue.[/quote]

ui thats a pretty cool feature :smile: … i’ve patched it now to 1.4.25
here is the trace :smile:

6

well, i’m not 100% sure what to tell you at this point. there are a couple of things I would try.

the first would be to add a record to /etc/hosts with a name for the IP of the proxy server. Then modify your sip.conf to reference the host name instead of the IP.

I have had to do this for a couple of systems which were multi-homed or behind NAT. Not sure this will work for you, but it’s worth a shot. I am heading out now, but i’ll check back in on this thread later.

7

[quote=“g2010”]well, i’m not 100% sure what to tell you at this point. there are a couple of things I would try.

the first would be to add a record to /etc/hosts with a name for the IP of the proxy server. Then modify your sip.conf to reference the host name instead of the IP.

I have had to do this for a couple of systems which were multi-homed or behind NAT. Not sure this will work for you, but it’s worth a shot. I am heading out now, but i’ll check back in on this thread later.[/quote]

ok, thx … i did it

added to the /eth/hosts

172.16.5.51    innovaphone

than change the sip.conf and try to add a global register

looks now that way:

and the registration works fine

put quite the same result :frowning:

i also took a trace with wireshark from the working softphone solution:

thx in advance for helping :smile:

8

ok solved the problem :smile:

this one helped me
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf