New install, cannot make a call out

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Hi,
I have just about finished Part I of the Orgasmatron II instllation and I cannot make a test call.

I have a standard Walmart special (Everex gPC2); Aastra 57i and 57i ct phones (one of each); and a D-link WBR-2310 Router with the wireless turned off (wired service only).

I have followed the instructions from the October 1, 2008 posting of The Asterisk Mother Lode; Introducing the Orgasmatron II for the $199 Everex gPC2. I have finished installing/setting up the phones and have restarted them.

When I tried to make a test call as outlined in the instructions, it was not completed. The phones all registered with the computer without a problem. The 57i ct base phone and wireless paired and have established an intercom connection. No intercom connection between either of the 57i ct sets and the 57i set.

All three phones have a dial tone. After dialing 1-2-3-4 to make the test call the phone no-longer has a dialtone and nothing happens. I cannot hear anything on the line. I have dialed 1-2-3-4 and waited over a minute for the call to be connected, but it does not connect.

I have tried to dial an eleven digit number (1 plus area code and number; which should allow me to register for the Nerd's call out service) and the same thing happens. When I press the first key the dialtone is turned off and the line is dead.

From the Everex box I can ping each of the phones, the router, the DSL modem and internet sites (both by IP and name).

During the installation of piaf I did not notice any problems.

Any help will be greatly appreciated.

Thanks-
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
OK, I am back. Thanks for all of the help on this problem.

I found two problems, one of which I have fixed and the other is still unresolved. The problem I fixed was updating the firmware on the phones. Now I get a busy signal and a message on the phone's display "call failed". I also have a message on the display that says "No Service".

The still unresolved problem is the phones keep changing their DNS server. The correct setting is nnn.nnn.1.1. But everytime I check the setting on the phones it is nnn.nnn.0.1. I keep changing the setting on the phones and when they re-start the setting is bad to the wrong one.

I have tried to place a SIP call to the server using the "mothership....." IP address the instructions say should work. Nothing happens on the server, but on the calling end of the call, I can hear it ring 4 times and hang up.

Tring to place a call to a 10 diget number also does not work.
 

jroper

Guru
Joined
Oct 20, 2007
Messages
3,832
Reaction score
71
Are you using DHCP in your network, or static IP addresses.

It does sound as if these are network problems you are having.

Joe
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Thanks for the reply Joe. I have the server on a static IP and the phones on DHCP IPs. The phones have IP's assigned correctly; the IP, Netmask and Gateway are correctly assigned, but the DNS is incorrect.
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Hi again Joe.

I just changed the settings on one phone to static so it would retain the DNS, but I still have the same problems:

1. The dislay on the phone shows a "No Service" message.

2. When I try to dial any number I get a massage "Call Failed" and a busy signal.

Since I do not have a VoIP carrier yet, this would make sense to me if the instructions did not say something else would happen.

Any suggestions?
 

jroper

Guru
Joined
Oct 20, 2007
Messages
3,832
Reaction score
71
Despite the fact you have not got a VoIP carrier, you have not got a SIP trunk set up have you, even a dummy one? A quick hunt round the forums will show that this is a problem if the SIP trunk is registered

Can you ping the phones from the PBX.

If you type set sip debug does it give you any clues.

Joe
 

tomsyr

Guru
Joined
Oct 26, 2007
Messages
266
Reaction score
1
Hi Autumn,
Are you able to call each phone on your network? Are the phones registered?
You can see if they are or not by going into FreePBX - then Tools, Asterisk Info and see how many sip peers you have. It should show at least the amount of phones you have.
For your phones getting DHCP addresses - do you have PC's that have this problem with DNS?

TomSyr
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
I can ping both phones by IP from the PBX. I can also ping both phones by IP from the computer I used to open the PBX and change the pass words. From both computers I can ping a site on the internet by either IP or domain name.

The phones have some kind of internet connection because they both up dated their firmware.

I had tried to call the PBX with the [email protected] address from another computer, but nothing happened on the PBX and the other computer hungup after 4 rings.

I have not setup a POTS on the PBX or a soft phone because I have the two Aastra phones ready to test and setup the outbound nd inbound trunks.

Typing set sip debug does not give me anything, it just returns to the Root prompt.
 

tomsyr

Guru
Joined
Oct 26, 2007
Messages
266
Reaction score
1
What extensions did you give the Aastra phones? If it's 201 and 202 for example, you should be able to use the 201 phone and call 202.
Did you take a look at Asterisk Info?
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Tom,

In the Asterisk info I have the following:

Active SIP channels: 0
SIP Registary: 1
SIP Peers: Online: 4
Offline: 16

Active IAX@ Channels: 1
IAX2 Registry: 0
IAX2 Peers: Online: 0
Offline: 1
Unmounted: 0

On this LAN I have the following devices connected:
PBX
Laptop used to access the FreePBX, router, etc.
1 Asstra 57i
1 Astra 57i CT with Wireless handset

The 57i, PBX, Laptop and router are using static settings and the 57i CT is using DHCP handed out by the router.
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Tom,

The 57i is extension 702 and the 57i CT is 701.

If I dial 702 on the 57i CT I get a quick flash of a message "Unknown number" and the "Call Failed" and the busy signal.

I get the same thing if I reverse and dial the 701on the 57i.

If I dial 1-2-3-4 on either phone I get a quick flash of a message "Unknown number" and the "Call Failed" and the busy signal.
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Hello Joe,

I was typing at the root prompt. Now I have changed to Asterisk - prompt pbx*CLI

But when I type set sip debug I get a message "No such command 'set sip debug'.

I do not have a SIP provider configured. Nor do I have a Dummy sip trunk con figured.

In reading the instructions (The Asterisk Mother Lode: Introducing the Orgasmatron II for the $199 Everex gPC2) in the section entitled "Adding Plain Old Phones" it says "Before your new PBX will be of much use ... Today , you've got seneral choices: the POTS phone, a softphone or a SIP phone (highly recommended)."

I took this to mean I could use any of all of the options. I chose the SIP phone option and did not bother with either of the others. I skipped right to the "Configuring Aastra 57i SIP Phones" and followed those instructions.

Are you suggesting that I must somehow configure a sip provider and/or a Dummy sip trunk before I setup the 57i phones? If that is what you are suggesting, where do I find the instructions to do this?

As you can tell I am new at all of this, but I cn follow instructions if I m pointed in the right direction. The last time I did annthing like this was 20+ years ago with a Nortel switch.

Thanks,
 

tomsyr

Guru
Joined
Oct 26, 2007
Messages
266
Reaction score
1
Autumn,
If you don't have a remote ssh client, you should install it on your laptop:
http://www.putty.org/
This way you can copy/paste the output from your session.
If or when you are set up with this, log on, then type in:
asterisk -rvvvvvv
then try to call ext. 702 with the 701 phone. Copy and paste the output so we can see how the call is being processed.
Here is a reference:
http://www.voip-info.org/wiki-Asterisk+CLI
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Tom, here is what happens when I try to call ext 702 from ext 701. Ext 702 has a static IP and 701 DHCP.
******************************
== Parsing '/etc/asterisk/asterisk.conf': Found
Connected to Asterisk 1.4.21.2 currently running on pbx (pid = 2872)
Verbosity is at least 6
pbx*CLI>
<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7ebd925aebeaf66c6
Max-Forwards: 70
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>
Call-ID: 83485a0bab156b9d
CSeq: 29992 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Ext-701 <sip:[email protected]:5060;transport=udp>
Supported: timer, 100rel, replaces
User-Agent: Aastra 57iCT/2.1.1.71
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.0.101
s=SIP Call
c=IN IP4 192.168.0.101
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:eek:n - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 25 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
Using INVITE request as basis request - 83485a0bab156b9d

<--- Reliably Transmitting (NAT) to 192.168.0.101:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7ebd925aebeaf66c6;received=192.168.0.101
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>;tag=as5e15a6a5
Call-ID: 83485a0bab156b9d
CSeq: 29992 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02d4f9c1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '83485a0bab156b9d' in 32000 ms (Method: INVITE)
Found user '701'
pbx*CLI>
<--- SIP read from 192.168.0.101:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7ebd925aebeaf66c6
Max-Forwards: 70
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>;tag=as5e15a6a5
Call-ID: 83485a0bab156b9d
CSeq: 29992 ACK
User-Agent: Aastra 57iCT/2.1.1.71
Content-Length: 0


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

<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7edabd74235dfdbc4
Proxy-Authorization: Digest username="701",realm="asterisk",nonce="02d4f9c1",uri="sip:[email protected]:5060",response="3d3c0bd511751984064e84a51bef553b",algorithm=MD5
Max-Forwards: 70
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>
Call-ID: 83485a0bab156b9d
CSeq: 29993 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Ext-701 <sip:[email protected]:5060;transport=udp>
Supported: timer, 100rel, replaces
User-Agent: Aastra 57iCT/2.1.1.71
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.0.101
s=SIP Call
c=IN IP4 192.168.0.101
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:eek:n - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 25 lines) ---
Sending to 192.168.0.101 : 5060 (NAT)
Using INVITE request as basis request - 83485a0bab156b9d
Found user '701'

<--- Reliably Transmitting (NAT) to 192.168.0.101:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7edabd74235dfdbc4;received=192.168.0.101
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>;tag=as5e15a6a5
Call-ID: 83485a0bab156b9d
CSeq: 29993 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '83485a0bab156b9d' in 32000 ms (Method: INVITE)
pbx*CLI>
<--- SIP read from 192.168.0.101:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7edabd74235dfdbc4
Max-Forwards: 70
From: Ext-701 <sip:[email protected]:5060>;tag=5af3410c8c
To: 702 <sip:[email protected]:5060>;tag=as5e15a6a5
Call-ID: 83485a0bab156b9d
CSeq: 29993 ACK
User-Agent: Aastra 57iCT/2.1.1.71
Content-Length: 0
 

jroper

Guru
Joined
Oct 20, 2007
Messages
3,832
Reaction score
71
Hi

Back to basics - but first a couple of corrections.

As you have worked out, the command for debugging SIP is sip set debug, my mistake.

Secondly, glad to hear that you do not have a dummy sip provider set up, that could have screwed things up.

The output shows that the call is not being authorised. This would suggest username and password issues.

I'd go back to basics, create an extension with a secure password, or secret.

Factory reset your phone to get rid of any misconfigurations.

Carefully type in the extension/username into the phone GUI and likewise the password secret. I don't have an aastra to hand, so I give you a step by step, but maybe someone who does can.

When you've done this, check asterisk info to check the phone is registered, and try *43 to make a echo test call.

Then try dialling extensions.

Joe
 

jroper

Guru
Joined
Oct 20, 2007
Messages
3,832
Reaction score
71
PS

Have you configured externip and localnet, what have you put in there.

Any other changes to the system from the standard install?

Joe
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Hi Joe,

In response to your most recent post; I have not configured externip and localnet. The only things that have been done to the phones other than the instructions in The Aterisk Mother Lode: Introdcing the Orgasmatron II for the $199 Everex gPC2 is to set one of the phones to static IP instead of the normal DHCP and the updating of the firmware on both phones. The firmware update was done automatically one time when they rebooted.

I have not had time to try your suggestions from your second most recent post (the resetting to factory settings, etc.), but I will do it asap and post the results.

Thanks,
 

tomsyr

Guru
Joined
Oct 26, 2007
Messages
266
Reaction score
1
I'm not sure why the following is in your output:
Sending to 192.168.0.101 : 5060 (NAT)
Using INVITE request as basis request - 83485a0bab156b9d
Found user '701'

<--- Reliably Transmitting (NAT) to 192.168.0.101:5060 --->
SIP/2.0 403 Forbidden

From what you have described in your setup, there shouldn't be any NAT (Network Address Translation) going on.

Like Joe has pointed out, reconfiguring your phones would be the best route to take.
 

autumn.tel

Member
Joined
Dec 14, 2008
Messages
38
Reaction score
0
Hello Joe and Tom,

I hope you both had a nice holiday.

Well, have been busy testing this setup and after many installs and much testing I have some:

Bad news- the problem still is there;
Good news- I know what causes the problem;
Bad news- I do not know how to fix the problem; and
Good news, I hope- you guys will have the answer.

Joe, I did what you suggested in your last post; I reset the phones to factory settings and did the setup on them. I had the same problem.

I decided to try a new install of PiaF because after all of the fooling around someting could by amiss.

After the new install (using the same CDs) I was carefull and checked everything I entered to be sure there were no mistakes. I went through the Tutorial step-by-step not changing or adding anything that was not in the tutorial. When I got to part in the tutorial where I should have been able to test the phones (call 1234, call one extension from the other, call 511, etc., etc.) I had the same problem. The phones displayed the "No Service" message and all calls failed.

Since I had used the same .iso CDs for all of the installs I had made, I thought maybe there was something wrong with the CDs. So, I downloaded another copy of PiaF from DreamHost and burned new CDs.

I tried to do an install using the new .iso CDs. The install froze at 96% complete. I rebooted and tried again. I got the freeze again at the 96% completion mark.

Back to DreamHost. I downloaded a third copy of PiaF from DreamHost and burned a third set of .iso CDs. I did another install. This time when I reached the point of restarting the phones I heard a beep and the display did not say "No Service"

I made the test calls (1234, 701 > 702, 702 . 701, *43, *60, *63, etc. Everything worked! At last I had success. I was ready to setup some trunks.

I dialed a ten digit phone number as the tutorial instructed. Nothing happened! I could not believe it. I tried again. Nothing. Itried to dial the number from the other phone. Again, nothing happened.

I thought for a while. I went over the tutorial step-by-step looking for something I may have missed. I opened PiaF. I went through the settings. It was then it struck me. When I was doing the setup someone caame in and took me away for a while. When I came back I did not properly finish the setup. Instead of starting where I had left off I jumped ahead. I skipped the steps of changing the 16 passwords and setting up DISA, DISAmain, PIN; of entering the FQDN information by editing the ddclient.conf file, and all that goes with it.

I went back and did all of that and it broke the system. When I finished the phones displayed the "No Service" message and all call atteempts failed. Back to where I started.

I thought about this for a long time. I decided the problem must be in the steps I had left out, so I should do another install leaving those steps out again. When I had the phones working I would do the left-out steps one-by-one and check the phones after each step.

Phone 701 stopped working when I used PiaF to reset the default password to a new password.

I have been able to duplicate this problem on three additional occassions.

Phone 702 continues to work as-long-as I do not reset the default password.

Do either of you have any suggestions on how to fix this problem?

Thanks,
 

Members online

Forum statistics

Threads
25,778
Messages
167,504
Members
19,198
Latest member
serhii
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Top