Asterisk Setup for Community VoIP

Thanks Ian, that's what I did.

Using the #81. prefix I noticed in the Asterisk debug file that the softphone (I use SJPhone but I also think it was the same with the Xten I tried) or Asterisk seemed to be replacing the # witha 3 digit numerical code for #(I think but did not keep a note of it) or something and sending this string to Asterisk. So stripping off the first 3 characters of the string gave a different number to the one I thought I was passing to Asterisk which led to a 404 error. Does that make sense?

I was just curious as to why my Asterisk setup was receiving a numeric code for # rather than the # symbol itself and whether that was a config issue in Asterisk or something to do with my set up.

If other noobies follow the original code with #81 they may well have the same problem as me and give up. It might be a good idea to replace the #81 in the original guide with a pure numerical prefix.

I would find it strange if I am the first person to to come across this problem.

Cheers

TonyB

Hi Tony

I have just edited an extensions.conf file to start with #6 for voipuser and it works fine from my deskphone and x-lite. I would have a look in the features.conf and see if anything is in there. What version of * are you using ?

Ian

The three figure code wouldn't be 035 would it? (# in ASCII)

Check that your keyboard is set up correctly and you haven't got anything remapping the keys.

Hi, I've rerun again with _8 and _#8 as dial prefixes and appropriate strip setups, the former works and the latter gives a 404 error and I get the following SJPhone log snippets (phone no replaced with number):

_8. Prefix

10:04:08 INFO Initiating SIP call to sip:8(number)@192.168.1.12
10:04:08 DEBUG
2005-10-21 09:04:08.907 UDP LOCAL->192.168.1.10:5060
INVITE sip:8(number)@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12;rport;branch=z9hG4bKc0a8010c000000524358af0800006a79000000fc
Content-Length: 337
Contact: <sip:201@192.168.1.12:5060>
Call-ID: A6BF7799-9487-4374-8DD7-F1EB04DEEBB0 [!at] 192.168.1.12 (replace the [!at] with a @)
Content-Type: application/sdp
CSeq: 1 INVITE
From: "Tony Batty"<sip:201@192.168.1.12>;tag=149341725155
Max-Forwards: 70
To: <sip:8(number)@192.168.1.12>
User-Agent: SJphone/1.60.289a (SJ Labs)

_#8 Prefix

09:58:57 INFO Initiating SIP call to sip:#8(number)@192.168.1.12
09:58:58 DEBUG
2005-10-21 08:58:58.090 UDP LOCAL->192.168.1.10:5060
INVITE sip:%238(number)@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12;rport;branch=z9hG4bKc0a8010c000000464358add1000029c3000000ca
Content-Length: 337
Contact: <sip:201@192.168.1.12:5060>
Call-ID: 0C4F0E19-910F-45BE-BAD0-D5F41D036965 [!at] 192.168.1.12 (replace the [!at] with a @)
Content-Type: application/sdp
CSeq: 1 INVITE
From: "Tony Batty"<sip:201@192.168.1.12>;tag=118255029061
Max-Forwards: 70
To: <sip:%238(number)@192.168.1.12>
User-Agent: SJphone/1.60.289a (SJ Labs)

At some point in the handshaking process the # gets changed into %23 (hexadecimal ascii code for #). My gut feeling is that this is neither a SJPhone or local machine issue but at the asterisk server end as I had similar problems with X ten on another local network machine.

I've also tried stripping 1-7 characters from the _#8. number all result in 404 errors. which probably suggest that the % is not allowed.

For information I am using the Xorcom distribution with new sip.conf and extensions.conf files.

TonyB

Hi Tony.

Ok This isnt an Asterisk bug in its self, Xorcom does use macros similar to AAH, This must be in the macro that this is happening as I have tested it on * and dont get any errors. I will be doing some testing on my lab system that is based on xorcom so I can see if it is a script problem. Personally startin a dial string with a hash is not good practice as # is the standard end of dial string for telecoms networks.

Ian

Thanks Ian. I don't think it is one of the Xorcom macros as essentially I've replaced the sip and extensions conf files with new simple conf files to help me learn Asterisk, so it shouldn't be calling any macros from within those files. Maybe there is conf file elsewhere in the Xorcom distribution that I need to check.

I used the Xorcom distribution because I am very much a Linux beginner (slightly better than newbie ) and the distribution seems ok for me with a menu driven linux frontend that helps. Having said that I do get a couple of error notifications. Are you a Xorcom expert? Which forum would you suggest I post the queries to? I also have a query on SJPhone also.

Thanks to the forum, I now have a working Voip User connection and am looking to extend that to Sipgate and beyond.

Thanks again for your help.

TonyB

Hi Tony

I have tried the #8 on my xorcom box and its fine with it using xlite, As I mentioned using a # may not be a good idea, and my reservation was well founded as you cant dial a number starting with a # from a Budgetone.

But here is the sip debug and its all aok

Quote:

Sip read: INVITE sip:#82206@192 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.65:5060;rport;branch=z9hG4bKA6300A25CD8D4DE986DA88DA51CE338D From: server <sip:502@192>;tag=3047436604 To: <sip:#82206@192> Contact: <sip:502@192.168.10.65:5060> Call-ID: 014EBE94-1421-4584-9037-121B8F40F552 [!at] 192.168.10.65 (replace the [!at] with a @) CSeq: 4319 INVITE Proxy-Authorization: Digest username="502",realm="asterisk",nonce="12889b7b",response="358c72ac27d9eb2faee98a0a68cd2fbd",uri="sip:#82206@192" Max-Forwards: 70 Content-Type: application/sdp ser-Agent: X-Lite Free World Dialup build 1082 Content-Length: 310

as to

Quote:

I used the Xorcom distribution because I am very much a Linux beginner (slightly better than newbie Smile ) and the distribution seems ok for me with a menu driven linux frontend that helps. Having said that I do get a couple of error notifications. Are you a Xorcom expert? Which forum would you suggest I post the queries to? I also have a query on SJPhone also.

Well post your errors here, I do at the moment have 2 xorcom boxes in the lab as well as the live vanilla one. I wont claim to be an expert (But then again I dont claim to be an expert in anything) but xorcom is so close to vanilla * debuging it is simple. as to sjphone post it in the sip software forum Im sure someone will help.
BTW xorcom do have a forum but when I posted a query I had with ISDN and TDM400 cards mixed I never got any help
But then again its a bit of an odd build with multiple isdn2 cards and a tdm400 for fxo.

Ian

Hi Dean

I'm trying to configure my Asterisk box for outbound calls.
I'm callin a friend in the UK.
Below you can see what I've got. Any idea?

First trial +44(798)470-xxxx :
-- Executing Dial("SIP/200200-2f24", "SIP/44798470xxxx@voipuser||Ttr") in new stack
-- Called 447984700318@voipuser
-- Got SIP response 404 "Not found" back from 83.143.18.16
-- SIP/voipuser-4780 is circuit-busy

Second trial +44(0798)470-xxxx :
-- Executing Dial("SIP/200200-13f9", "SIP/440798470xxxx@voipuser||Ttr") in new stack
-- Called 4407984700318@voipuser
-- Got SIP response 404 "Not found" back from 83.143.18.16
-- SIP/voipuser-e3c7 is circuit-busy

sip.conf
========
[voipuser]
type=friend
context=inbound_voipuser
username=fgomes
secret=xxxxxxxx
host=voipuser.org
fromuser=fgomes
fromdomain=voipuser.org
insecure=very
qualify=yes


Thanks!

Fernando Gomes

Hi

You have two problems,

1. Your trying to call a mobile which wont work.
2. The dial string is just the UK number with no 44 so for example to call a london number it would be 0201231223 not 442021231223

Hope that helps Ian

Hi there,

I have compiled and installed the latest version of the asterisk server
on a Linux(centos) box. I am now trying to configure outgoing dialing using
a voipuser account. Unfortunately after trying out the procedure described
on the webpage :
"asterisk support for community VoIp) at : http://www.voipuser.org/forum_topic_330.html

but I am unfortunately unable to make outgoing calls. I get a
call failed (user not found) error. I am using the latest version of
XTEN lite as my SIP client. My Aterisk installation otherwise
works and I have verified this by calling from one XTEN
phone to the other XTEN client successfully.

I have also independently configured XTEN to use my voipuser account
information directly and I was able to successfully call a phone
number in the US.

I am copying below the additions to sip.conf and extensions.conf
(changed IP address to 100.200.300.x , username etc). These machines
are not behind a NAT and directly connected to the internet. I am
also attaching below the SIP messages by turning on "sip debug" in
the CLI console of Asterisk.

Any help would be appreciated !
thanks
Yuvraj


--------------------------------
Inserted in sip.conf
--------------------------------
;insert by Username - start
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw
allow=all
;inserted by Username – finish


;Inserted by Username - Start
register => username:password@voipuser.org/448449335187

[voipuser]
type=friend
context=incoming_voipuser
username=username
secret=password
host=voipuser.org
fromuser=username
fromdomain=voipuser.org
insecure=very
qualify=no
disallow=all
allow=g729
allow=gsm
allow=ulaw

;Inserted by Username – Finish

--------------------------------
Inserted below in extensions.conf
--------------------------------
;Inserted by Username - Start voipuser account stuff
[incoming_voipuser]
exten => 448449335187,1,NoOp(--- ${CALLERID} calling on VoIPUser (${EXTEN}) ---)
exten => 448449335187,2,Dial(SIP/yoursipphone,20)
exten => 448449335187,3,Answer
exten => 448449335187,4,Wait,1
exten => 448449335187,5,Voicemail(u1)
exten => 448449335187,6,HangUp
;To make calls through voipuser.org by dialing 999<number to call>:
[outgoing_voipuser]
exten => _999.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
exten => _999.,2,Congestion
;Inserted by Username - Finish

--------------------------------
Inserted below the log from "sip debug on" inside asterisk
--------------------------------


--------------------------------------------------<-- SIP read from 100.200.300.245:41616:
INVITE sip:9990018581234567@100.200.300.243 SIP/2.0
Via: SIP/2.0/UDP 100.200.300.245:41616;branch=z9hG4bK-d87543-c53bd3408c474d78-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:username@100.200.300.245:41616>
To: "9990018581234567"<sip:9990018581234567@100.200.300.243>
From: "Username (dar)"<sip:username@100.200.300.243>;tag=485f2909
Call-ID: 3e7d90208c635803MjFlYWU0NTI1YjdkMmNmZjE0MThkNDc3ZjU1NzA0NjY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1003l stamp 30942
Content-Length: 385
v=0
o=- 7 2 IN IP4 100.200.300.245
s=CounterPath eyeBeam 1.5
c=IN IP4 100.200.300.245
t=0 0
m=audio 45750 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : IEpFDGgo MTlbl1iu 100.200.300.245 45750
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:BDDBB5576D9F4D3E972A53ED54D8FC14

--- (12 headers 14 lines)---
Using INVITE request as basis request - 3e7d90208c635803MjFlYWU0NTI1YjdkMmNmZjE0MThkNDc3ZjU1NzA0NjY.
Sending to 100.200.300.245 : 41616 (NAT)
Found user 'username'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 100.200.300.245:45750
Found description format BV32
Found description format BV32-FEC
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 9990018581234567 in dar-users (domain 100.200.300.243)
Reliably Transmitting (no NAT) to 100.200.300.245:41616:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 100.200.300.245:41616;branch=z9hG4bK-d87543-c53bd3408c474d78-1--d87543-;rport;received=100.200.300.245
From: "Username (dar)"<sip:username@100.200.300.243>;tag=485f2909
To: "9990018581234567"<sip:9990018581234567@100.200.300.243>;tag=as01810731
Call-ID: 3e7d90208c635803MjFlYWU0NTI1YjdkMmNmZjE0MThkNDc3ZjU1NzA0NjY.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990018581234567@100.200.300.243>
Content-Length: 0


---

<-- SIP read from 100.200.300.245:41616:
ACK sip:9990018581234567@100.200.300.243 SIP/2.0
Via: SIP/2.0/UDP 100.200.300.245:41616;branch=z9hG4bK-d87543-c53bd3408c474d78-1--d87543-;rport
To: "9990018581234567"<sip:9990018581234567@100.200.300.243>;tag=as01810731
From: "Username (dar)"<sip:username@100.200.300.243>;tag=485f2909
Call-ID: 3e7d90208c635803MjFlYWU0NTI1YjdkMmNmZjE0MThkNDc3ZjU1NzA0NjY.
CSeq: 1 ACK
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '3e7d90208c635803MjFlYWU0NTI1YjdkMmNmZjE0MThkNDc3ZjU1NzA0NjY.'

<-- SIP read from 100.200.300.245:41616:



--- (0 headers 1 lines)---

<-- SIP read from 100.200.300.2:62616:

Ok, I followed the above instructions and I see this:

Sip Peers
Host Username Refresh State
voipuser.org:5060 ***** 120 Auth. Sent


I never seems to register? what am I doing wrong?

Hi.

I've been looking the forum for several days and I can't find a solution for my problem. I'll try to explain it clearly:

I've got to make calls through asterisk using my voipuser account, but I cannot receive calls (using my voipuser number). So I think asterisk is working correctly, it must be a configuration problem.

The device I'm using is a x-lite softphone on the same pc than asterisk, it registers and everything's ok when I make calls, but for some reason, it is not detecting the incoming calls.

This computer is behing NAT, and i think i've configured it properly. These are the relevant lines of extensions.conf:

Code:

[from-sip] ;eco exten => 500,1,Answer() exten => 500,2,Echo() exten => 500,3,Hangup() exten => 600,1,Answer() exten => 600,2,Dial(SIP/9990,20) exten => 600,3,Hangup() ;include => outgoing_calls include => outgoing_voipuser ;recibir llamadas de voipuser exten => 08449792372,1,NoOp() exten => 08449792372,2,Dial(SIP/9990,20) exten => 08449792372,3,Answer exten => 08449792372,4,HangUp exten => 448449792372,1,NoOp() exten => 448449792372,2,Dial(SIP/9990,20) exten => 448449792372,3,Answer exten => 448449792372,4,HangUp exten => 00448449792372,1,NoOp() exten => 00448449792372,2,Dial(SIP/9990,20) exten => 00448449792372,3,Answer exten => 00448449792372,4,HangUp exten => _X.,1,NoOp("call for "${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@voipuser,60,tr) exten => _X.,n,Hangup()

And these are from sip.conf:

Code:

[general] ;mis codecs disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw allow=all context=default    bindport=5060    bindaddr=0.0.0.0    srvlookup=yes                          ; Configuracion por estar bajo NAT nat=yes externip=xxx.xxx.xxx.xxx localnet=192.168.222.1/255.255.255.0 canreinvite=no ;Registro en voipuser register => USER:PASS@voipuser.org/NUMBER ; Cuenta de voipuser [voipuser] type=friend context=incoming_voipuser username=USER secret=PASS host=voipuser.org fromuser=USER fromdomain=voipuser.org insecure=very qualify=no ;canreinvite=no disallow=all allow=g729 allow=gsm nat=yes externip=XXX.XXX.XXX.XXX localnet=192.168.222.1/255.255.255.0 ; dos clientes [9990] type=friend username=9990 secret=palabra host=dynamic context=from-sip mailbox=9990 nat=no canreinvite=no [9999] type=friend username=9999 secret=palabra host=dynamic context=from-sip mailbox=9999 nat=no canreinvite=no

What am I doing wrong?
How can I know if Asterisk is detecting the incoming calls from voipuser?
Is asterisk registering in a proper way to voipuser?

Please, any idea will be apreciated.

Thank you very much

PD: Forgive my english.

Thank you very much

HI,
i tried your configuration on my asterisk pbx it is working fine with single username and password.
but i want to register more than one voipuser on my pbx.
how.
status :
the voipuser is same as your configuration.
i made voipuser1 with diffrent userid on same host
incoming calls coming properly.
but circuit busy in outgoing call
here i am using sip-outgoing for ist service provider
and sip-outgoing1 for second service provider.
the dial pattern are as follow

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

[outgoing_voipuser1]
exten => _#91.,1,Dial(SIP/${EXTEN:3}@voipuser1,60)
exten => _#91.,2,Congestion

the cli output is
-- Executing [#811008@outgoing_voipuser:1] Dial("Zap/1-1", "SIP/1008@voipusk -- Called 1008@voipuser
[Dec 31 16:20:23] WARNING[199]: chan_sip.c:11860 handle_response_invite: Receiv' -- SIP/voipuser-0080375c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [#811008@outgoing_voipuser:2] Congestion("Zap/1-1", "") in newk == Spawn extension (outgoing_voipuser, #811008, 2) exited non-zero on 'Zap/1-' -- Hungup 'Zap/1-1'

-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/4-1'
-- Executing [#911008@outgoing_voipuser1:1] Dial("Zap/4-1", "SIP/1008@voipuk -- Called 1008@voipuser1
-- SIP/voipuser1-00b44004 is ringing
== Spawn extension (outgoing_voipuser1, #911008, 1) exited non-zero on 'Zap/4' -- Hungup 'Zap/4-1'

it means only one voipuser is working successfully.
whichever register at last.

Thanks
Roopesh Motewar

Only one Voipuser account is allowed, you will have to set up with a commercial provider for your second and subsequent lines.

Thanks gray,
Actually, i am running asterisk as sip provider, asterisk pbx as a sip client.
what should i do on server side so it will work fine.
presently i am using only extensions setting to register it.
present status is 2 voipusers are registered successfully.
one of them is full duplex other one is half duplex.

Thanks
Roopesh