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Today's Topics:

   1. Re: PRI channels suspended (Ashish gautam)
   2. Re: IVR, bridge, ring back problem (Paul Cupis)
   3. Re: Newbie question about inbound calls/NAT (yes I checked
      wiki first:-) ) (Cal Leeming [Simplicity Media Ltd])


---------- Forwarded message ----------
From: Ashish gautam <ashish@nms.co.in>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Tue, 16 Apr 2013 14:46:58 +0530
Subject: Re: [Freeswitch-users] PRI channels suspended

2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended

This is what happening on restarting/starting the FS for all the channels(this is only for channel number 30).

How to resolve it? Please help.

--

Regards.

On Tue, Apr 16, 2013 at 2:46 PM, Ashish gautam <ashish@nms.co.in> wrote:

Sorry Michael mistakenly sent to you personally again. Sending to the list once more.

On Tue, Apr 16, 2013 at 2:45 PM, Ashish gautam <ashish@nms.co.in> wrote:

2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended

This is what happening on restarting/starting the FS for all the channels(this is only for channel number 30).

How to resolve it? Please help.

--

Regards.

On Mon, Apr 15, 2013 at 9:51 PM, Michael Collins <msc@freeswitch.org> wrote:

Please only send this to the list, otherwise it is nearly impossible for me to reply in such a way that everyone on the list can join the conversation.
-MC

On Sun, Apr 14, 2013 at 11:29 PM, Ashish gautam <ashish@nms.co.in> wrote:

Hi,

I am having another problem over here. When I run "ftdm dump 1 5" from fs_cli I get my channel status as SUSPENDED. This happens for channel number 5 to 30. Also when I try to make outgoing calls more than 4 in number, four calls originate properly whereas for the rest it shows NORMAL_CIRCUIT_CONGESTION. Here is the output:

span_id: 1

chan_id: 5

physical_span_id: 1

physical_chan_id: 5

physical_status: ok

physical_status_red: 0

physical_status_yellow: 0

physical_status_rai: 0

physical_status_blue: 0

physical_status_ais: 0

physical_status_general: 0

signaling_status: UP

type: B

state: SUSPENDED

last_state: RESTART

txgain: 0.00

rxgain: 0.00

cid_date:

cid_name:

cid_num:

ani:

aniII:

dnis:

rdnis:

cause: NORMAL_UNSPECIFIED

session: (none)

-- States --                   -- Function --                 -- Location --                 -- Time Offset --

DOWN  => RESTART               [on_dchan_up]                  [ftmod_libpri.c:2372]          0ms

RESTART => SUSPENDED           [on_timeout_t316]              [ftmod_libpri.c:1952]          120008ms

Time since last state change: 324790932ms

Please help me getting out of this.

Thanks.

--Ashish

--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

---------- Forwarded message ----------
From: Paul Cupis <paul@cupis.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Tue, 16 Apr 2013 10:54:32 +0100
Subject: Re: [Freeswitch-users] IVR, bridge, ring back problem
On Sat, Apr 13, 2013 at 07:12:58AM +0200, Andrzej wrote:
> I'll check on Monday.
> but:
> 1. Is this normal behavior IVR?

This is normal behavior, yes - the call has already been answered, so the
caller is not going to be sent 180 Ringing or 183 Early media in order to
hear ringing during the bridge - you need to provide audio to fill the
silence.

> 2. How it should be written:

a. with a space, not a colon.

Regards,





---------- Forwarded message ----------
From: "Cal Leeming [Simplicity Media Ltd]" <cal.leeming@simplicitymedialtd.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Tue, 16 Apr 2013 11:43:15 +0100
Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) )

Glad to hear you got it sorted, and thank you for posting back your finding - hopefully this will help others in the future!

Cal

On Tue, Apr 16, 2013 at 9:29 AM, Jeff Bernhardt <jeff@askcornerstone.net> wrote:

You know, I actually just figured it out myself just a couple hours ago.... but I'm gonna give your method a shot too since I haven't played with ACLs yet and that would give me a chance to.

I discovered that the problem was actually how I'd setup Vitelity to route to the server. I didn't realize this before, but in Vitelity, you can either route to the server IP address to allow IP based authentication OR you can route to the account login name that the trunk is being registered with. The former was giving errors since I didn't have that account setup, whereas the latter worked with no problem. I'm guessing this is because with the IP based auth method, Vitelity connects to port 5060 (can't change this), which is taken as Freeswitch's default internal profile port (requires authentication) instead of port 5080 (the public profile with doesn't require authentication?), and since there was no internal user for what Vitelity was sending essentially unsolicited on 5060, it got denied. As to why it works by routing to the registered account login name I'm not sure, but perhaps that's part of what registration does; it sends to the public profile by default...? 

Anyway, thanks for all the help! Great to see such an active and helpful list with something I'll need a lot of help with!

---

From: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] on behalf of Michael Collins [msc@freeswitch.org]
Sent: Monday, April 15, 2013 7:15 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) )

For inbound calls you can skip the authentication by adding the source IP address to the "domains" section of conf/autoload_configs/acl.conf.xml. This will prevent FreeSWITCH from challenging the inbound call if it comes from that IP address. This is more desirable than disabling call authentication altogether in your SIP profile. (i.e. you are allowing only a specific IP address to come in unauth'd as opposed to letting the whole world in!)

Find out the source IP address for your Vitelity SIP traffic and add it to the acl.conf.xml file in the "domains" section. It will be something like this:

<node type="allow" cidr="1.2.3.4/32"/>

After you save that go to fs_cli and type "reloadacl". NOTE: simply doing a "reloadxml" is not sufficient to make the new ACL entry take effect. ("reloadxml" simply reloads the XML config; "reloadacl" does a reloadxml and then reloads the ACL entries.)

Hope this helps.
-MC

On Mon, Apr 15, 2013 at 12:30 AM, Jeff Bernhardt <jeff@askcornerstone.net> wrote:

Thanks. I had a look at the links and changed "domain and even "domain_name" in vars.xml to be the external ip, but the error now just says  it can's find user abcd_1.2.3.4@1.2.3.4 instead of @192.168.10.32.

And actually, I 'm mostly confused because I thought incoming calls were treated as unauthenticated and allowed to pass through. Isn't registering the trunk with the SIP provider enough to get incoming calls accepted? 

The errors says "...you must configure your device to use the proper domain in it's authentication credentials," but how is the SIP trunk treated as a device?

Like I said, I think I'm misunderstanding something fundamental about this.

Thanks you.

---

From: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] on behalf of Cal Leeming [Simplicity Media Ltd] [cal.leeming@simplicitymedialtd.co.uk]
Sent: Sunday, April 14, 2013 3:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT (yes I checked wiki first:-) )

Hello Jeff,

Here is a related thread;

http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-called-xx-com-in-your-directory-td7585944.html

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232.html

You need to configure your domain and username, but as you can see here your gateway/users are sending the domain '192.168.10.32', but you have not configured this domain in sofia.conf.

If you are using multi-tenant then you'll need to put some more thought into how you're going to approach this.

If you are not using multi-tenant (i.e. one domain only) then you could force the domain as discussed here;

http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freeswitch-problems/

Hope this helps

Cal

On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt <jeff@askcornerstone.net> wrote:

Setting up my first test box with Freeswitch. I'm really excited about getting it up and running.

I'm trying to get inbound calls working from external using a Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and password is "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get registered to Vitelity fine  (their web panel shows I'm registered and fs_cli confirms). However, when I make a call inbound, I get:

[WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4@192.168.10.32] from <vitelity-inbound-trunk-ip>

You must define a domain called '192.168.10.32' in your directory and add a user with the id="abcd_1.2.3.4" attribute

and you must configure your device to use the proper domain in it's authentication credentials. 

I don't know if this is related, but to get around NAT, I have this in vars.xml:

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>

<X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>

and this in both internal.xml and external.xml:

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<param name="ext-sip-ip" value="$${external_sip_ip}"/>

I have a feeling I've misunderstood something fundamental about setting up Freeswitch for inbound access. I've put in the above configs as per what I've found in the wiki, but I'm not sure how to do what the warning says or if it really means I have something not setup right. What am I doing wrong?

Thank you.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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