Dial out using Asterisk almost there!

Hi Ian

Setup outgoing as you suggested.

After dropping the firewall (That can come later ! )
I get the dialed number ringing, but that is all.

This is the setup: I have two phones 2002 & 2004. 2002 is where incoming is redirected. In theory if I dial my voipusers number from 2004 is should ring 2002 - but it does no.
If I dial my voipuser number from a telco phone, then it work 100%

This is what I get on the console"

Quote:

*CLI> -- Accepting AUTHENTICATED call from 213.166.5.129, requested format = 8, actual format = 4 -- Executing Dial("IAX2/serik@213.166.5.129:4569/2", "sip/2002|20|r") in new stack -- Called 2002 -- SIP/2002-d533 is ringing == Spawn extension (incoming_voipuser, 448700689890, 1) exited non-zero on 'IAX2/serik@213.166.5.129:4569/2' -- Hungup 'IAX2/serik@213.166.5.129:4569/2

Quote:

*CLI> -- Executing SetCallerID("SIP/2004-bb23", "Serik") in new stack -- Executing Dial("SIP/2004-bb23", "IAX2/voipuser.org/08700689890|60|r") in new stack -- Called voipuser.org/08700689890 Jan 27 21:08:05 WARNING[147465]: chan_iax2.c:1473 attempt_transmit: Max retries exceeded to host 216.127.66.119 on IAX2/216.127.66.119:4569/4 (type = 6, subclass = 1, ts=10, seqno=0) -- Hungup 'IAX2/216.127.66.119:4569/4' == No one is available to answer at this time -- Executing Congestion("SIP/2004-bb23", "") in new stack == Spawn extension (from-sip-internal, 4408700689890, 3) exited non-zero on 'SIP/2004-bb23' -- Executing Hangup("SIP/2004-bb23", "") in new stack == Spawn extension (from-sip-internal, h, 1) exited non-zero on 'SIP/2004-bb23' DIALED FROM 2004

After the prescriber period ===> Congestion

Here is the extensions.conf snippet

Quote:

;[voipuser-out] exten => _44.,1,SetCallerId,Serik exten => _44.,2,Dial(IAX2/voipuser.org/${EXTEN:2},60,r) exten => _44.,3,Congestion

Can you see why it is not working?

Kind regards
Serik

Serik,

Destinations on the outbound have a limit in what they can cost, basically anything around 2p is fine but anything more then that (a minute) is not allowed.

So in short you can't dail the 0870, 0871 and 070 numbers on the free outbound.

Mind you, you can dail the My SIP number of another member as they are routed internally thru the SIP network.

Regards,

Tj

Hi,

serik :

Can you see why it is not working?

Yes -

serik :

exten => _44.,2,Dial(IAX2/voipuser.org/${EXTEN:2},60,r)

They don't do outgoing over IAX - only SIP. See my thread here for setup instructions: http://www.voipuser.org/forum_topic_330.html

--ian

Tj

Quote:

Mind you, you can dail the My SIP number of another member as they are routed internally thru the SIP network.

This is what I thought I was doing - keeping the call "Internal" and not incur an cost, and still allow me to run various test etc.

Looking at the table of various numbers 0844, 0870 etc., the lowest cost is the 0844 numbers. Will these allow me to do this?

Why is the 0844 number rate cheaper, and can I use it? The "outgoing number allocated to me is 0844xxxxxxx

Serik

serik :

Why is the 0844 number rate cheaper, and can I use it? The "outgoing number allocated to me is 0844xxxxxxx

The 0844 allocated with the outbound account is a 3p/min 0844 but you can only use this with the sip account - you can't terminate it anywhere else.

The 0844 in "My Numbers" is a 5p/min 0844 and the termination destination can be set - because there is more revenue so they can afford different terminations options.

--ian

Hi Ian

Right! Ive changed the outgoing to SIP

Extensions.conf is

Quote:

exten => _44X.,1,Dial(SIP/${EXTEN:3}@serik,60) ;exten => _44.,2,Dial(IAX2/voipuser.org/${EXTEN:2},60,r) exten => _44X.,3,Congestion

Console gives

Quote:

-- Executing Dial("SIP/2002-e2c6", "SIP/1536521717@serik|60") in new stack Jan 28 20:20:25 NOTICE[262160]: app_dial.c:742 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing Hangup("SIP/2002-e2c6", "") in new stack == Spawn extension (from-sip-internal, h, 1) exited non-zero on 'SIP/2002-e2c6'

It looks as if I am loosing the "0" even though I am dialing it!
I have tried various formats like
"_44.Dial(Sip etc"
"_44X.Dial(Sip etc"
"_#44.Dial(Sip etc"
but cannot get the "0"

I am sure this is why it is not getting through.

Serik

Do you know ${EXTEN:3} removes the 1st 3 digits?

I use:
exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
..to force calls through voipuser - so I dial: #8101234123123 and it sends 01234123123 to voipuser.

I also use:
exten => _0[1-6].,1,Dial(SIP/${EXTEN}@voipuser,60)
..for uk national numbers (so I dial 01234123123)

and:
exten => _00.,1,Dial(SIP/${EXTEN}@voipuser,60)
..for international numbers (so I dial 00...........)

--ian

Thanks Ian!!

Quote:

Do you know ${EXTEN:3} removes the 1st 3 digits?

Oooops! Yes! I did not see that when I copy/pasted. I've now changed ${EXTEN:3} to ${EXTEN:2} and it is dialing out right.

I have dropped the firewall and tried to call a UK number rings (But I cannot heard the ringing). It gets answered OK but no audio. CODEC? used ulaw and iLbc.

Any suggestions?

Regards
Serik

Quote:

It gets answered OK but no audio.

Port forwarding?

Dean

HI Dean

SIP - 5060
Iax - 4569
Router (ADSL) has DMZ set to forward to Asterisk Server where the firewall is as well..

Do I need any other ports?

But I know I got a problem with the firewall/NAT which we are working on, so when I need to test, I drop the firewall, do what I want to test, then bring up the firewall again.

Serik

serik :

SIP - 5060 Iax - 4569 Do I need any other ports?

Yes - the rtp ports from /etc/asterisk/rtp.conf.

--ian

serik :

CODEC? used ulaw and iLbc.

By the way, it's worth noting that ilbc works on the incoming IAX termination but not on sip - incoming or outgoing.

I use g729 with the sip outgoing and it's incoming 0844 number (had to buy licences from Digium at $10 each to use g729 with Asterisk) and ilbc for IAX incoming.

--ian