Cannot place call to teliax, "user not found"?

Asterisk Asterisk Support
1

I can call my asterisk server from a land line, but I cannot make an outgoing call from a softphone to a land line.
The softphone says, “user not found”.
Teliax has tripped the switch to allow authentication to be in the body of the pasket.
Still doesn’t work.

Here is my extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
PSTN=Zap/g2

[default]
exten => 8005181896,1,Answer()
exten => 8005181896,2,Playback(dir-intro)
exten => 8005181896,3,Queue(service|t||8005181896|45)

[outbound]
exten => 8008629121,1,Answer()
exten => 8008629121,2,Playback(demo-congrats)
exten => 8008629121,3,AgentLogin()

exten => h,1,DeadAGI(postqueue.agi)

[8008629121]
;exten => 8008629121,1,Answer()
;exten => 8008629121,1,DIAL(SIP/user,20)

[204]
exten => _1XXXXXXXXXX,2,DIAL,(IAX2/xxxx@teliax/${EXTEN},30,tr)
exten => 204,3,Answer
exten => 204,4,Hangup

here is my sip.conf

register => xxxx:xxxxxxxxxxx@voip-co3.teliax.com
[authentication]
auth = xxxxx:xxxxxxxxxxxx@voip-co3.teliax.com

[teliax]
context=default
type=friend
username=xxxxx
user=xxxxx
host=voip-co3.teliax.com
secret=xxxxxxxxxxxxx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[204]
user=204
context=internal
type=friend
secret=xxxxxxxxx
insecure=very
canreinvite=no
context=home
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=ilbc ; preference
allow=gsm
nat=no

Here is the debug

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.176.193.46:5072;branch=z9hG4bKf8452f47-cdf8-1810-869e-0013d3ee21fe;received=66.176.193.46;rport=5072
From: “Brad S” sip:204@xx.xx.xx.xx;tag=74272f47-cdf8-1810-869a-0013d3ee21fe
To: sip:19544790554@xx.xx.xx.xx;tag=as7878bf48
Call-ID: 59212f47-cdf8-1810-869a-0013d3ee21fe@usmc
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘59212f47-cdf8-1810-869a-0013d3ee21fe@usmc’ in 32000 ms (Method: INVITE)

<— SIP read from 66.176.193.46:5072 —>
ACK sip:19544790554@xx.xx.xx.xx SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP 66.176.193.46:5072;branch=z9hG4bKf8452f47-cdf8-1810-869e-0013d3ee21fe;rport
From: “Brad S” sip:204@xx.xx.xx.xx;tag=74272f47-cdf8-1810-869a-0013d3ee21fe
Call-ID: 59212f47-cdf8-1810-869a-0013d3ee21fe@usmc
To: sip:19544790554@xx.xx.xx.xx;tag=as7878bf48
Proxy-Authorization: Digest username=“204”, realm=“asterisk”, nonce=“59963977”, uri="sip:19544790554@66.109.17.92", algorithm=md5, response="08af1ad02b83d5a1c8a4fb442588d9ea"
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Length: 0
Max-Forwards: 70

I can see the it is putting my extension @ my_sip_server, but it should be looking at the body of the message right now.

How do I make it put the user account in the header instead of the extension?

2

Can someone recommend a good free soft phone?

I am thinking this is a Ekiga issue!!!

Thanks!

3

Ok, I install Adorees soft phone and I have the same issue.
Here is the overview of everything, I am at a complete and total loss.

I can call into my asterisk server from a land line and listen to the welcome message, but no SIP phones can make outbound calls.

I have the Teliax recommended setting but still nothing.

Can some kind person please take a look and tell me where I am going wrong?

Teliax recommends this:

[general]
register => xxxx:xxxxxxxxxxx@voip-co3.teliax.com
And add the following context to your sip.conf -

[authentication]
auth = xxxx:xxxxxxxxxxx@voip-co3.teliax.com

[teliax]
context=default (or a valid context in your extensions.conf of your choosing.)
type=friend
username=UXMC
user=UXMC
host=voip-co3.teliax.com
secret=xxxxxxxxx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

(you may set as many codecs to “allow” as you like. We will configure your connection according to the chosen
settings from your account at www.teliax.com.)

STEP 2

Using the service:

The following is an example for use in the asterisk extensions.conf file that should allow you to terminate calls (dial out).

exten => _1XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},30,tr)

And the following line is an example of what is needed for TelIAX originated calls (incoming calls). This example assumes you
are using an SIP phone but an SIP phone is, of course, not required.

exten => YOURNUMBER,1,Answer()
exten => YOURNUMBER,1,DIAL(SIP/user,20)

I have the following in sip.conf

register => xxxxxxxxx:xxxxxxxxxxxx@voip-co3.teliax.com
[authentication]
auth = xxxx:xxxxxxxxxx@voip-co3.teliax.com

[teliax]
context=default
type=friend
username=UXMC
user=UXMC
host=voip-co3.teliax.com
secret=xxxxxxxxxxx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[UXMC]
user=xxxx
context=internal
type=friend
secret=xxxxxxxxxxxx
insecure=very
canreinvite=no
context=home
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=ilbc ; preference
allow=gsm
nat=no

I have the following in extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
PSTN=Zap/g2

[default]
;exten => 8005181896,1,Answer()
;exten => 8005181896,2,Playback(dir-intro)
;exten => 8005181896,3,Queue(service|t||8005181896|45)

[outbound]
exten => 8008629121,1,Answer()
exten => 8008629121,2,Playback(demo-congrats)
exten => h,1,DeadAGI(postqueue.agi)

[8008629121]
;exten => 8008629121,1,Answer()
;exten => 8008629121,1,DIAL(SIP/user,20)

[UXMC]
exten => _1XXXXXXXXXX,1,DIAL,(SIP/teliax/${EXTEN},30,tr)
exten => 8005181896,1,Answer
exten => 8005181896,2,Hangup

And here is my debug from an attempted call

<— SIP read from 66.176.193.46:2190 —>
INVITE sip:19544790554@66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 1 INVITE
Contact: sip:66.176.193.46:11214
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 448

v=0
o=- 0 0 IN IP4 66.176.193.46
s=session
c=IN IP4 66.176.193.46
b=CT:1000
t=0 0
m=audio 46340 RTP/AVP 97 111 112 6 0 8 4 5 3 101
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
— (11 headers 20 lines) —
Sending to 66.176.193.46 : 11214 (no NAT)
Using INVITE request as basis request - 36ab7cec5c6542c191d64520253925a5@66.176.193.46

<— Reliably Transmitting (no NAT) to 66.176.193.46:11214 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92;tag=as0b374235
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17d55c85"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘36ab7cec5c6542c191d64520253925a5@66.176.193.46’ in 32000 ms (Method: INVITE)
Found user ‘UXMC’

<— SIP read from 66.176.193.46:2190 —>
ACK sip:19544790554@66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92;tag=as0b374235
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 1 ACK
User-Agent: RTC/1.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from 66.176.193.46:2190 —>
INVITE sip:19544790554@66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 2 INVITE
Contact: sip:66.176.193.46:11214
User-Agent: RTC/1.2
Proxy-Authorization: Digest username=“UXMC”, realm=“asterisk”, algorithm=MD5, uri="sip:19544790554@66.109.17.92", nonce=“17d55c85”, response="06a3dd76f64fa529c2f87842a6adfdec"
Content-Type: application/sdp
Content-Length: 448

v=0
o=- 0 0 IN IP4 66.176.193.46
s=session
c=IN IP4 66.176.193.46
b=CT:1000
t=0 0
m=audio 46340 RTP/AVP 97 111 112 6 0 8 4 5 3 101
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
— (12 headers 20 lines) —
Sending to 66.176.193.46 : 11214 (no NAT)
Using INVITE request as basis request - 36ab7cec5c6542c191d64520253925a5@66.176.193.46
Found user 'UXMC’
Found RTP audio format 97
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 66.176.193.46:46340
Found description format red for ID 97
Found description format SIREN for ID 111
Found description format G7221 for ID 112
Found description format DVI4 for ID 6
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format DVI4 for ID 5
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0xc3f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc|g726aal2)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.176.193.46:46340
Looking for 19544790554 in home (domain 66.109.17.92)

<— Reliably Transmitting (no NAT) to 66.176.193.46:11214 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92;tag=as0b374235
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘36ab7cec5c6542c191d64520253925a5@66.176.193.46’ in 32000 ms (Method: INVITE)

<— SIP read from 66.176.193.46:2190 —>
ACK sip:19544790554@66.109.17.92 SIP/2.0
Via: SIP/2.0/UDP 66.176.193.46:11214
Max-Forwards: 70
From: “UXMC” sip:UXMC@66.109.17.92;tag=cad85e11a8654f28ab8cf63a785fd2c4;epid=1114cb07d4
To: sip:19544790554@66.109.17.92;tag=as0b374235
Call-ID: 36ab7cec5c6542c191d64520253925a5@66.176.193.46
CSeq: 2 ACK
User-Agent: RTC/1.2
Content-Length: 0

4

Perhaps you should try making it work with an IAX trunk instead, although inbound may be a problem, depending on your configuration.

5

Assuming the iaxtel.com website was up, I would register and try.
But it is down.

I read earlier about trying teliax as an aix trunk and configure accordingly.
Note my earlier postings, “user not found”…

This is blowing my mind and really ticking me off that every single piece of lituature I fin bounces to a hundred different places and never makes a relevent point.

Everyone has the same beginner issue.
Relationship between globals / sip.conf registration / user authentication ADD THEN HAVE THE POLITE CONSIDERATION to carry it over to exentions.conf for a basic inbound / outbound call.

The viop.org files are just frustrating me and ol Marks Handbook is about worthless too.

My apologies to the list for venting my friends!

6

Here is a novel concept!
Does anyone know where I can get a copy of AMP without Asterisk@home?

How about the millions of people whom lease dedicated servers and do not or want the ability to boot up from a disk?

I have command line access only and there seems to be little support of zero way to get the GUI only!

7

SOMEBODY HAS GOT TO SAY SOMETHING, so I guess it is going to be me.

Click on ‘Setup’ at the top of the page, then click on ‘Trunks’ on the left, Then click ‘Add IAX2 Trunk’. (THERE IS NO SETUP BUTTON ON THE COMMAND LINE, CAN YOU GIVE ME THIS INFO FOR A CLI SETUP?)

Enter the following information in the appropriate fields:

Outbound Caller ID: “Your-Name” (WHICH FILE ARE YOU REFERING TO? sip.conf, extensions.conf, or iax.conf???)

Maximum Channels: 4 (varies from plan to plan. residential plans include 2. corporate plans include 4.) (WHAT CHANNELS, WHERE, WHAT FILE, WHO HOW AND WHAT???)
Dial Rules:

1|NXXNXXXXXX <----- This will allow you to call 1+AreaCode+Number
(NOW THIS APPEAR NOW WHERE LIKE THIS BUT HERE I AM ASSUMMING extensions.conf??? I checked the asterisk handbook 3 times. CAN YOU ELABORATE???)
NXXNXXXXXX <----- This will allow you to call just AreaCode+Number
316+NXXXXXX <----- This will allow you to dial 7 Digit numbers for the 316 areacode.
Change the areacode to match your own areacode to dial 7
Digits from your home phone.

Trunk Name: teliax
Peer Details:
(NOW COMMON SENSE SAYS TO PUT THIS IN iax.conf, but IT WOULD BE NICE IF YOU SPECIFIED!!!)
allow=ulaw
auth=md5
context=default
disallow=all
host=YOUR-PROXY <----- This can be found near top of support page within Teliax Portal.
nat=yes
secret=YOUR-PASSWORD <----- This can be found near top of support page within Teliax Portal.
type=friend
username=YOUR-USERNAME <----- This can be found near top of support page within Teliax Portal.

User Context: teliax-in
User Details:
(USER DETAILS IN sip.conf???)

allow=ulaw
auth=md5
context=from-pstn
disallow=all
type=friend

Registration String: YOUR-USERNAME:YOUR-PASSWORD@YOUR-PROXY

Configure Inbound Routing for each Teliax DID

In AMP (Asterisk Management Portal), click on ‘Setup’ at the top of the page, then click on ‘Inbound Routing’ on the left, then click ‘Add Incoming Route’.

DID Number: 3165551212 (replace with your DID)
Set Destination: Use ‘Incoming Calls’ Setting

Configure Outbound Routing for calls through Teliax

In AMP (Asterisk Management Portal), click on ‘Setup’ at the top of the page, then click on ‘Outbound Routing’ on the left, then click ‘Add Route’.

Route Name: teliax
Dial Patterns:

1NXXNXXXXXX <----- This will match 1+AreaCode+Number
NXXNXXXXXX <----- This will match AreaCode+Number
NXXXXXX <----- This will match local seven-digit Number

Trunk Sequence: IAX2/teliax

8

HA!
AMP does not supprt 1.4!

All this conversation about 1.2 EOL

So this puts me at square one!

Geeze!

How about some CLI support!

9

If you have a Teliax account, you can use IAX. Go into your account and make sure the IAX box is ticked.

After looking over your configuration again, I notice the following:

If you don’t have your authentication info in your iax2.conf, that won’t work.

Your later configuration looks like it should work, AFAIK.

I just tested SIP with them and it works fine as long as I have the register statement in the [general] section of sip.conf. If I don’t register, they reject the call attempt.

I only use IAX for trunks, so I can’t be of more assistance than to say try IAX, since teliax supports it.

10

go to freepbx.org. you can run AMP without running trixbox.

11

I went to the Teliax support site and did the IAX configuration to the letter.
Nothing
Zip
zero

Same problem

12

I did the freepbx install without trixbox

that is when install script told me that AMP does not support Asterisk 1.4
AMP will support in ver 2.3

So, as of yesterday
NO SUPPORT FOR AMP since Asterisk 1.2 is End of Life!

So, technically, AMP is dead in the water till they release AMP 2.3

13

working now

Had to delete an existing and working config completely to get it too work.