Topic: As soon when picked up the phone disconnects :/

[wifi]

Hello.. Me again..

I thought I had it all figured out..
I now have two trunklines setup that both register fine with the providers.
One is Voipbuster, and the other is Budgetphone.

Both accounts have a phone in number, and both numbers / accounts work great when calling out to any phone number in the workd.
The Voipbuster account is routed to extension 5000 (a Grandstream BT 100 IP phone), and working perfect (in and out)
The Budgetphone account is routed to extension 5001 (SJphone), and when called it rings, BUT when answered (picked up) it disconnects again straight away.

As test I "replaced" the SJphone with the Grandstream phone, and the same thing happens. As soon as I pickup the phone it disconnects the calling person (like putting the phone down again)

The route to the phone is okay (it rings) and the trunklines are okay (they are both registered).
Now what else could this be?

[SARK devs]

You'll need to post a copy of the console log when the call comes in.

log in to linux and then into asterisk using asterisk -rvvvv

run your call and grab the log.

Best

S

[wifi]

Here you go (I hope I'm not posting anything sensitive)

I have masked the real number (last 7 digits) with xxxxxxx
I have also removed the working VB account from the server (I will set it up again later)
Shown here is the bad account used with ext: 5000 (Grandstream BT 100).
Calling out no problem. Receiving the call also not problem.. Answering WILL drop the line/

I used the info found here: http://www.budgetphone.nl/forum/viewtopic.php?t=51 to setup the trunk for Budgetphone
The 1st post on that forum is for a working Asterisk setup.

[root@files ~]# asterisk -rvvvv
Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.18 currently running on files (pid = 4458)
Verbosity was 0 and is now 4
    -- Executing [3130xxxxxxx@mainmenu:1] AGI("SIP/3130xxxxxxx-0a1f2548", "selintra|Inbound|3130xxxxxxx") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [5000@internal:1] AGI("SIP/3130xxxxxxx-0a1f2548", "selintra|OutCluster|5000") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [5000@default:1] AGI("SIP/3130xxxxxxx-0a1f2548", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/5000||t)
    -- Called 5000
    -- SIP/5000-0a1f64c0 is ringing
    -- SIP/5000-0a1f64c0 answered SIP/3130xxxxxxx-0a1f2548
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/3130xxxxxxx-0a1f2548' status is 'ANSWER'
    -- Executing [h@default:1] Hangup("SIP/3130xxxxxxx-0a1f2548", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/3130xxxxxxx-0a1f2548'
[Mar 21 19:37:45] WARNING[4649]: chan_sip.c:1947 retrans_pkt: Maximum retries exceeded on transmission 31c26f946c6c5abd5a63875566fba54e@gw02-mci.budgetphone.nl for seqno 102 (Critical Response)
files*CLI>

[SARK devs]

This looks perfectly normal (at least from asterisk's point of view).  Here is a call being answered and clearing down on one of our test servers...(it looks just the same as yours except we cleared down normally from the sending phone.

Code: [Select]

    -- AGI Script selintra completed, returning 0
    -- Executing [4008@internal:1] AGI("Zap/1-1", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/4008|10|t)
    -- Called 4008
    -- SIP/4008-08a5e940 is ringing
    -- SIP/4008-08a5e940 answered Zap/1-1
    -- AGI Script selintra completed, returning -1
  == Auto fallthrough, channel 'Zap/1-1' status is 'ANSWER'
    -- Executing [h@internal:1] Hangup("Zap/1-1", "") in new stack
  == Spawn extension (internal, h, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'

Something is driving a hangup event into asterisk.  Run a SIP trace to verify this...

install ethereal if you don't already have it

Code: [Select]

yum install wireshark

Then run

Code: [Select]

tethereal -R sip -f  "host the-budgettone-url"
Now make your inound call and look for the hangup (BYE)  request being sent.

If it is the carrier then you'll need to speak with them.

[wifi]

Hi again,

It's doing nothing when calling and answering the number.

[root@files ~]# tethereal -R sip -f  "host sip.budgetphone.nl"
Running as user "root" and group "root". This could be dangerous.
Capturing on bond0

I guess sip.budgetphone is not correct. I've read something about this some time ago on the providers forum.
Let's see if I can find that info again.

[wifi]

Okay.

After changing a lot of things, I've managed to get it to work (for now)

The main change that got it working was in the sip.conf file (that I changed in Headers > sip.conf)
Normally it reads the following (localnet will be different)

[general]
tos=0x18
localnet=10.0.0.0/255.255.255.0
context=mainmenu
maxexpirey=180
defaultexpirey=160

After I changed it to what is shown below, calls can be received with the account (now I'm having problems making a call, but this could be due to some other changes that I made):

[general]
context=mainmenu
bindport=5060
bindaddr=0.0.0.0
svrloookup=yes

I will be making small modifications now to find the exact stuff that is needed and not needed.

[wifi]

Update 2.

After changing everything back to 'normal', and ONLY removing the localnet=10.0.0.0/255.255.255.0 from the Headers > sip.conf file, all is working fine (calling and being called)

I have no clue why localnet=10.0.0.0/255.255.255.0 is causing a problem for this SIP.
My LAN is within the 10.0.0.0/255.255.255.0 range!

I'm not sure by removing things like this how secure the setup is, but I guess time (or Selintra) will tell.


Anyway. Now I'm off adding the good old VB account back again.

Selintra. Thank you for all your help. I'll be back again.