Topic: Need help with getting my system working

[SmartSteve]

Hi All, Sorry if this appears elsewhere, i did have a good look but am in a state of frustration. I have a SME7.4 box that has been running fine, mostly used for web and email/mlist services. Recently I decided to ditch the landline I so faithfully paid each month but rarely used except for STD calls and went naked ADSL2. All that is running fine. The problem lies in my VoIP setup, which I only got just before I moved house and thus didn't have a chance to set up until now. Hardware is Linksys SPA3102 (yes i don't need the pstn side now, oh well), with a previously working phone plugged in. VoIP provider is Pennytel (recommended to me)

Initially I set up the spa3102 using the settings provided by PT and I could see the device registered on the PT website. I can ring my DID and the CID shows on my phone BUT you cannot answer the call and the calling (mobile) continues to ring. Also I cannot call OUT I just get a tone that I think means busy or something. From here I poured a lot of time into why such behaviour was happening. Research told me there were many ways to skin the cat and I tried several STUN servers and looked at siproxd (won't compile) BUT many people say the best way was set up a asterisk server (and config with sail, couldn't get freepbx to install). At least sail is what is shown in the manual (http://sarkpbx.com/twiki/bin/view/Main/DocChapter253). Which I followed faithfully.

At present I have (linksys) showing it is registered to 192.168.1.4 (SME/ast) and in asterisk -r I see the box connect with success (ext5000). I have also installed x-lite (ext5001) so i can have two options to test with and between.

In sail I see the trunk to pennytel is connected (little grey things together) and ACT has a little grey tick. In the extensions I see both spa3102 and x-lite are connected with the relevant IP(s) being correct. Also in asterisk -r I see both of the extensions connect. Which all looks good.

Now if I ring out I get the same beep beep beep that I got before but in asterisk -r i see
[May 24 12:19:55] ERROR[9371]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe (by 20 times).
If I ring in I get a voice (from pennytel) that say "that account does not exist" which is the same as if I don't have the spa3102 connected and the "normal" sip settings from pennytel. Also the PT website doesn't show any connection from asterisk.

I get the same broken pipe if I try to call ext5000 from 5001 and vice versa. I have all call recording set to NONE

So I am at my wits end. I have a wife who is at me every day as to why the phone is not working.

Setup is asterisk14 (1.4.30) from ATRPMS (was getting no where trying to compile the sources) with SAIL 2.2.4-44 from localrpm.
I want all the phone calls in and out to go via the VoIP provider.

That is about all I can think of to put down.

Cheers,

Steve B

[chris burnat]

Same problem here installing on fresh 7.5RC2 system.
I used Idkeen method and got a hitch:
yum --enablerepo=atrpms install dahdi-linux dahdi-linux-kmdl-`uname -r`
[No Match for argument: dahdi-linux-kmdl-2.6.9-89.0.23.EL]

[root@sail7b5 ~]# uname -a
Linux sail7b5 2.6.9-89.0.23.EL #1 Wed Mar 17 06:22:05 EDT 2010 i686 i686 i386 GNU/Linux

So went to  http://wolfcastle.wu-wien.ac.at:
wget http://wolfcastle.wu-wien.ac.at/dist/rhel4-i386/RPMS.at-stable/dahdi-linux-kmdl-2.6.9-89.0.23.EL-2.3.0-68.el4.i686.rpm
wget http://wolfcastle.wu-wien.ac.at/dist/rhel4-i386/RPMS.at-stable/dahdi-linux-2.3.0-68.el4.i386.rpm

And done:
yum localinstall dahdi-linux-kmdl-2.6.9-89.0.23.EL-2.3.0-68.el4.i686.rpm dahdi-linux-2.3.0-68.el4.i386.rpm
Installed: dahdi-linux.i386 0:2.3.0-68.el4 dahdi-linux-kmdl-2.6.9-89.0.23.EL.i686 0:2.3.0-68.el4

[root@sail7b5 sail]# rpm -q asterisk14 sail dahdi-linux dahdi-linux-kmdl-`uname -r`
asterisk14-1.4.30-90.el4
sail-2.4.1-23
dahdi-linux-2.3.0-68.el4
dahdi-linux-kmdl-2.6.9-89.0.23.EL-2.3.0-68.el4

yum list | grep dahdi
dahdi-linux.i386                         2.3.0-68.el4           installed       
dahdi-linux-kmdl-2.6.9-89.0.23.EL.i686   2.3.0-68.el4           installed       
dahdi-tools.i386                         1:2.3.0-66.el4         installed     

I got one extension registered, and a trunk up and ready.

CLI:
Asterisk 1.4.30, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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 1.4.30 currently running on sail7b5 (pid = 5684)
Verbosity is at least 5


Test call:
   -- Executing [0298xxxxxx@internal:1] AGI("SIP/5001-00000002", "selintra|OutCluster|0298xxxxxx") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/selintra
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
  ==  selintra|OutCluster|0298184139: Failed to execute '/usr/share/asterisk/agi-bin/selintra': No such file or directory
[May 25 08:37:01] ERROR[6253]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
  == Auto fallthrough, channel 'SIP/5001-00000002' status is 'UNKNOWN'
    -- Executing [h@internal:1] Hangup("SIP/5001-00000002", "") in new stack
  == Spawn extension (internal, h, 1) exited non-zero on 'SIP/5001-00000002'

[chris burnat]

Just got this error again on a test machine. The codec was set to Fidelity in General settings. Changing this to Thruput fixed this issue. Please do not ask why....