Asterisk Setup for Community VoIP

Hi ~
After many hours still having problems registering with voipuser.org. Our account is set up (I believe). We're behind a firewall-NAT, 5060 goes straight through to our * box. We've followed the setup carefully, but we're getting 'Unauthorized' replies from voipuser.org. Here's the packets we see on the firewall. Really appreciate some (any!) advice:

Code:

We said: 11:01:44.873327 216.46.157.177.53444 > 216.127.66.119.sip:  [udp sum ok] udp 377 (DF) (ttl 63, id 8596) REGISTER sip:voipuser.org SIP/2.0 Via: SIP/2.0/UDP 216.46.157.177:5060;branch=z9hG4 From: <sip:cxtinac@voipuser.org>;tag=as6e3f11bc To: <sip:cxtinac@voipuser.org> Call-ID: 1f57274b2d65d565299f184950e5f4bf [!at] 127.0.0.1 (replace the [!at] with a @) CSeq: 106 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:448449865628@216.46.157.177> Event: registration Content-Length:0 Voipuser.org said: 11:01:45.047074 216.127.66.119.sip > 216.46.157.177.sip:  udp 625 (DF) (ttl 48, id 11707) SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 216.46.157.177:5060;branch=z9hG4bK4e8c9dff From: <sip:cxtinac@voipuser.org>;tag=as6e3f11bc To: <sip:cxtinac@voipuser.org>;tag=38ea85d1dc04dd8ab8603fe6b0504472.5128 Call-ID: 1f57274b2d65d565299f184950e5f4bf [!at] 127.0.0.1 (replace the [!at] with a @) CSeq: 106 REGISTER WWW-Authenticate: Digest realm="voipuser.org", nonce="423316abdd4b367d77241ebdf1e24088f944bca4" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 216.12 01f0: 372e 3636 2e31 3139 3a35 3036            7.66.119:506

Any suggestions or help greatly appreciated...

Christina.

Hi,

cxtinac :

SIP/2.0 401 Unauthorized

Looks like your username and/or password is not correct.

Are you using the password on your voipuser.org account at the time of registering for the outbound access? - the change password feature doesn't change the sip password.

--ian

Hi ~ thanks for the reply! Yes, I am (unfortunately). To start with we weren't, but corrected that as part of the tries.

One other thing, a 'host' on our public IP fails, would voipuser.org require a good reply to this before an AUTH? e.g.

Code:

$> host pppoe.dyn-xxx.xxx.xxx.xxx.hurontel.on.ca $> Host pppoe.dyn-xxx.xxx.xxx.xxx.hurontel.on.ca not found: 3(NXDOMAIN)

Christina.

Hi ~
Well, we have finally managed to register successfully. Two things fixed it:

1. Our firewall was sending registration requests on a high port number -- it has to stay on 5060 through the f-w we found -- for us that means specifying "static-port".

2. We are registered using our updated password, not the original one. Has this changed maybe? (It is certainly working... or maybe I misunderstood...).

Thanks for all replies.
Christina.

When i try tio make an outbound PSTN call from a phone connected to my asterisk server (both inside NAT), I get only one way audio. I am on dynamic IP, so if I set externip in SIP, what happens when my IP changes? How can i solve this problem?

thx,
chuks

Using a hostname in sip.conf as:

Code:

externip=xxxxxxx.dnsalias.net ; Address that we're going to put in outbound SIP messages

-seems to work for us...
(and then using dyndns.org or similar obviously)

flying_walrus :

Code: -- Called 0016504750863 [!at] voipuser.org (replace the [!at] with a @) Jan 12 04:44:59 NOTICE[135385088]: chan_sip.c:6807 handle_response: Failed to authenticate on INVITE to '"Ben Calvert" <sip:Unknown@64.81.53.18>;tag=as74f3f6de'   == Spawn extension (outgoing, 4750863, 1) exited non-zero on 'SIP/lorax-11c9'  -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

I got the same error until I received an e-mail from voipuser.com saying that my PSTN account was activated - which came a bout 3 days after registering.

Hi,

I'm a complete n00b at * and Linux actually, but I'm really interested in this project and think I'd use the incoming mode more than the outgoing, but anyway I cant seem to get it working.

I'm using SJPhone connected to *, and it all works fine for internal services, but I cant seem to get VoIPuser stuff working.

Here is the exact (-passwords & -phone number) copy of the sip.conf and then extentions.conf... can anyone help?

SIP.CONF:

Code:

; Note: If your SIP devices are behind a NAT and your Asterisk ;  server isn't, try adding "nat=1" to each peer definition to ;  solve translation problems. [general] nat=yes externip=***.***.***.*** <= My external IP (edited) localnet=192.168.0.34/255.255.255.0 canreinvite=no disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw allow=all register => Phiebs:PASSWORD@voipuser.org/NUMBER (44xxxxx...) [voipuser] type=friend context=incoming_voipuser username=Phiebs secret=PASSWORD host=voipuser.org fromuser=Phiebs fromdomain=voipuser.org insecure=very qualify=no disallow=all allow=g729 allow=gsm #include sip_nat.conf #include sip_additional.conf

Last bit of extentions.conf

Code:

[incoming_voipuser] exten => MYNUMBER,1,NoOp(--- ${CALLERID} calling on VoIPUser (${EXTEN}) ---) exten => MYNUMBER,2,Dial(SIP/yoursipphone,20) exten => MYNUMBER,3,Answer exten => MYNUMBER,4,Wait,1 exten => MYNUMBER,5,Voicemail(u1) exten => MYNUMBER,6,HangUp [outgoing_voipuser] exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60) exten => _#81.,2,Congestion

Any help would be appreciated!!

Mat

p.s. By the way SJPhone just says "Call rejected - 404 not found"[/code]

Configged it, when I put in '#81014377647**' for the number to dial (#81 being the prefix and ** being the last two digits of my home phone number) i get 499: Not Acceptable here. I can receive incoming calls on my sip account number, but no audio goes, either way. I forwarded an 0871number to it an 871 number to it, and still no audio either way.

Code:

May 14 12:16:10 DEBUG[1415]: Stopping retransmission on '486E1938-4341-4B8F-A6C2-E79D8E0071DD@192.168.0.2' of Response 3485: Found May 14 12:16:10 DEBUG[1415]: Setting NAT on RTP to 0 May 14 12:16:10 NOTICE[1415]: No compatible codecs! May 14 12:17:02 DEBUG[1415]: Manager received command 'Command' May 14 12:17:02 DEBUG[1415]: Manager received command 'Command' May 14 12:17:52 DEBUG[1415]: Setting NAT on RTP to 0 May 14 12:17:52 DEBUG[1415]: Stopping retransmission on '1D0C85E7-A75C-49F3-B0ED-11703B90217C@192.168.0.2' of Response 38696: Found May 14 12:17:52 DEBUG[1415]: Setting NAT on RTP to 0 May 14 12:17:52 NOTICE[1415]: No compatible codecs! May 14 12:19:02 DEBUG[1415]: Manager received command 'Command' May 14 12:19:02 DEBUG[1415]: Manager received command 'Command' May 14 12:19:02 DEBUG[1415]: Manager received command 'Command'

Also, none of the service numbers work (*45 etc)

Not getting any audio passing through. Sending apparently is fine but no receive.

Im getting

Quote:

Jun 20 15:10:14 NOTICE[30882]: chan_sip.c:4052 sip_reg_timeout: -- Registration for 'kizmet@voipuser.org' timed out, trying again Jun 20 15:10:15 NOTICE[30882]: chan_sip.c:6836 handle_response: Failed to authenticate on REGISTER to '<sip:kizmet@voipuser.org>;tag=as0fbeb76a' Jun 20 15:10:34 NOTICE[30882]: chan_sip.c:4052 sip_reg_timeout: -- Registration for 'kizmet@voipuser.org' timed out, trying again Jun 20 15:10:35 NOTICE[30882]: chan_sip.c:6836 handle_response: Failed to authenticate on REGISTER to '<sip:kizmet@voipuser.org>;tag=as04ab38b8' Jun 20 15:10:54 NOTICE[30882]: chan_sip.c:4052 sip_reg_timeout: -- Registration for 'kizmet@voipuser.org' timed out, trying again Jun 20 15:10:55 NOTICE[30882]: chan_sip.c:6836 handle_response: Failed to authenticate on REGISTER to '<sip:kizmet@voipuser.org>;tag=as6baa98bf'

my sip.conf entry is:

Quote:

register => kizmet:PASSWORD@voipuser.org [voipuser] type=friend context=incoming username=kizmet secret=PASSWORD host=voipuser.org fromuser=PASSWORD fromdomain=voipuser.org insecure=very qualify=no

Im not sure whats wrong. i have reset my password a couple of times but that makes no difference.
this machine is a DMZ behind a ADSL connection.

any help appreciated.
.// Kizmet

Even though I got the mail that my outbound service has been activated, I cannot place any outgoing calls at all through voipuser.org

My account registers fine and I allow all codecs, so I don't really know what might be the issue, would anyone in UK care to try to call my voipuser UK number to see if incoming calls work?

08449863741

I don't know if you need to make any special settings at your end when I am behind nat, but it works fine with other sip providers....

Greetings from Sweden, Carl "ZaphodB" Andersson

ichilton :

flying_walrus : Jan 12 04:44:59 NOTICE[135385088]: chan_sip.c:6807 handle_response: Failed to authenticate on INVITE to '"Ben Calvert" <sip:Unknown@64.81.53.18>;tag=as74f3f6de' == Spawn extension (outgoing, 4750863, 1) exited non-zero on 'SIP/lorax-11c9' -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119 That's the exact errors that stephen was reporting and he just came back saying he had an incorrect password. Are you sure your password is correct?

I spent most of yesterday evening trying to get outbound calls working, getting the 'Failed to authenticate on INVITE' and '483 "Too Many Hops"' messages as above.

Eventually, I got it working, and I've experimented a bit more tonight.

In case this is useful to anyone else in the same position, I thought it might be worth posting what I found, because I couldn't find this solution in any of the searches I did. (Of course, I may just not be reading carefully enough!)


A bit of background ... I have an ADSL line coming in to a Draytek Vigor 2600VGi modem/router (running a hardware firewall with NAT), that's connected to a Red Hat Linux system with 2 Ethernet cards that is running an iptables firewall (with NAT). The Draytek has the relevant ports opened up and forwarded to the linux firewall.

I have Asterisk running on the firewall Linux system, and I'm trying to dial out through voipuser from an X-lite softphone running on a laptop running Windows which is on the secure side of the firewall.


My sip.conf (stripped of comments and stuff not relevant to this posting) is:

Code:

[general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no register => DavidJames:******@sip.voipuser.org/44844986**** externip = ***.***.org.uk localnet=***.***.***.0/255.255.255.0 localnet=192.168.0.0/255.255.0.0 [voipuser_sip] disallow=all allow=ulaw type=peer context=voipuser_sip username=DavidJames           ;-------------------------(1) fromuser=DavidJames           ;-------------------------(2) secret=****** host=voipuser.org fromdomain=voipuser.org insecure=very qualify=no nat=yes canreinvite=no [tcs10] type=friend regexten=20 username=tcs10                           ;--------------(3) callerid="David James on tcs10" (6010)   ;--------------(4) host=dynamic defaultip=193.195.132.10 dtmfmode=rfc2833 mailbox=9999@default context=sip nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=ulaw allow=alaw

The relevant line in extensions.conf was:

Code:

[***] exten => _**448.,1,Dial,SIP/${EXTEN:5}@voipuser_sip,tr

Dialling **44801xxxxxxxxx (my home number) on the X-lite, I was getting the two errors.

Turning on SIP debugging, I eventually discovered that the 'From:' line was

From: "callerid from the x-lite context" <sip:fromuser from the voipuser context@voipuser.org>

If there is no callerid in the x-lite context, then instead this line contains the username from the x-lite context.

The 'Proxy-Authorization:' line was

Proxy-Authorization: Digest username="username from the voipuser context" ...


To get rid of the errors seems to require that (a) the callerid in the 'From:' line, (b) the username in the 'From:' line, and (c) the username in the 'Proxy-Authorization:' line all be my voipuser username (DavidJames).

The first two are easy enough to set in the voipuser context (lines (1) and (2) in sip.conf).

If I set the callerid at line(4) or comment out the callerid at line (4) and set the username at line (3) then the outgoing call works.

However, I don't really want to set the callerid for the x-lite to my voipusername, so I ended up by using SetCallerId (line (5)) in the dialplan in extensions.conf, as follows:

Code:

[***] exten => _**448.,1,SetCallerid("DavidJames")           ;---(5) exten => _**448.,2,Dial,SIP/${EXTEN:5}@voipuser_sip,tr

Then everything works nicely.


The only thing that still puzzles me is that the twiki says (at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID) "Be aware that setting fromuser= in sip.conf will overide SetCallerID!"

Hi, I've been trying to get Asterisk up and running with Voipuser for the last week or so using the code in the original post and struggling to dial out, getting error 404 although I was pretty sure I had the right user name and password. Eventually after messing about I found that the problem for me at least was the use of the # symbol. Removing the # symbol and altering the Dial appropriately I found it worked !!

Is there something I need to set in a config file to allow Asterisk to use the # key (and maybe *?)symbol as part of a dial string? Or is this something unique to my set up?

Cheers

Hi

Code:

[outgoing_voipuser] exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60) exten => _#81.,2,Congestion

You can replace the #81 with anything you want.
just notr that the 3 after EXTEN is the number of digits stripped

so if for example you swap #81 for just 8 change the 3 for 1

Ian