VoIP Mailing List Archives

[Mark.Tabron at rnid-ty...]

Hi,

My first post to the list. I’m a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn’t quite correct.

Recently had a 9 channel ISDN30 (euro – q931) installed by BT (UK). We’ve hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.

The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)

Here are the confs / dialplan as I have them:

[span wanpipe]
name => OpenZAP
number => 1
trunk_type => e1
b-channel => 1:1-9
d-channel => 1:16

<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<pri_spans>
<span id="1">
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>

<include>
<extension name="outgoing-pri">
<condition field="destination_number" expression="^9(\d{7,15})$">
<action application="bridge" data="openzap/1/a/$1"/>
</condition>
</extension>
</include>

Do the config files look ok? Is there anything else I should be configuring? Is there anything else I can use to debug or get information on the PRI status?

Thanks in advance.
Mark.
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[msc at freeswitch.org]

[Mark.Tabron at rnid-ty...]

I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian).

Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer!

http://pastebin.freeswitch.org/7751

Will setup an IRC client and see if I can log onto the channel.

Thanks again!

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 12 March 2009 16:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

[Mark.Tabron at rnid-ty...]

Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room.

So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be.

Any ideas? Pastebin debug output is in my reply below.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron
Sent: 13 March 2009 14:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian).

Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer!

http://pastebin.freeswitch.org/7751

Will setup an IRC client and see if I can log onto the channel.

Thanks again!

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 12 March 2009 16:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

[msc at freeswitch.org]

Any chance you can give one of us access to this system? Best thing to
do would be to join #openzap on irc.freenode.net.
-MC (IRC: mercutioviz)

On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[Mark.Tabron at rnid-ty...]

Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though.

I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it!

Mark.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 16 March 2009 17:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

Any chance you can give one of us access to this system? Best thing to
do would be to join #openzap on irc.freenode.net.
-MC (IRC: mercutioviz)

On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[Mark.Tabron at rnid-ty...]

Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.

However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron
Sent: 17 March 2009 09:31
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though.

I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it!

Mark.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 16 March 2009 17:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

Any chance you can give one of us access to this system? Best thing to
do would be to join #openzap on irc.freenode.net.
-MC (IRC: mercutioviz)

On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[msc at freeswitch.org]

On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[Mark.Tabron at rnid-ty...]

We're a couple more steps forward from yesterday. Turned out some of my regex
was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra
space before one of the closing brackets in the default.xml example. After
staring at the screen all day it's funny how you miss these things!

Situation now is I can get the call into FS but, it rings the extension for
a fraction of a second then the call drops. Here's the contents of the
public and default dialplans I'm using (as per example in the wiki) and the
debug - http://pastebin.freeswitch.org/7819
http://pastebin.freeswitch.org/7819

I'm also seeing another issue when placing subsequent inbound calls, they
bounce if hitting the same channel the first call came in to (typically
/1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818
http://pastebin.freeswitch.org/7818

Getting there (slowly)

Mark.


mercutioviz wrote:

[Mark.Tabron at rnid-ty...]

We're a couple more steps forward from yesterday. Turned out some of my
regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has
an extra space before one of the closing brackets in the default.xml
example. After staring at the screen all day it's funny how you miss
these things!

Situation now is I can get the call into FS but, it rings the extension
for a fraction of a second then the call drops. Here's the contents of
the public and default dialplans I'm using (as per example in the wiki)
and the debug - http://pastebin.freeswitch.org/7819

I'm also seeing another issue when placing subsequent inbound calls,
they bounce if hitting the same channel the first call came in to
(typically /1:1). Again, grabbed a debug of this -
http://pastebin.freeswitch.org/7818

Getting there (slowly)

Mark.

<quote author="mercutioviz">
On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[Prometheus001 at gmx.net]

2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch
there was a timer problem which was not solved yet. This caused channels
to be busy in my case.

I am not sure whether this is solved yet. Can anybody confirm?

Best regards
Peter

Mark Tabron schrieb:

[msc at freeswitch.org]

On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX <Prometheus001@gmx.net> wrote:

[Mark.Tabron at rnid-ty...]

So the second issue is possibly known - really could do with a fix or a
workaround for this as we plan to use E1's for all incoming traffic.

Can anyone shed light on the first problem (extension rings for a
fraction of a second then hangs up) I mentioned below, or is that
possibly part of the same issue?

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
Peter P GMX
Sent: 18 March 2009 17:18
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch
there was a timer problem which was not solved yet. This caused channels
to be busy in my case.

I am not sure whether this is solved yet. Can anybody confirm?

Best regards
Peter

Mark Tabron schrieb:

[msc at freeswitch.org]

On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote:

[Mark.Tabron at rnid-ty...]

Thanks, found an install guide on the FS Wiki for libpri - will get the
server cloned then install and test.

Shall report back.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 19 March 2009 16:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem dialing out via E1

On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron
<Mark.Tabron@rnid-typetalk.org.uk> wrote: