[Freeswitch-users] FreeSWITCH-users Digest, Vol 82, Issue 133
Akash Deep Verma
akashdeep.co at gmail.com
Tue Apr 16 21:16:01 MSD 2013
Hi Ashish,
This is a hack but you can get what you need.
This is because, whenever retry attempt is made 1 channel per pri get
started, as there are 30 channels and retry attempts are 3 so the channels
that freeswitch load is in between 3-6, so I increased retry attempt and
complied it again. As I got rid of this error but it takes some time to
load the pri channels (30*10 = 300seconds(5 seconds) max).
Also, remember if you dial a call in between restarting phase you may face
some error. To avoid this, dial call only after all pri channel get started.
Here are the steps:
In freeswitch source:
Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h
1. Decrease timeout of channels.
#define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */
change this to
#define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */
2. Increase attempts to 30.
#define T316_ATTEMPT_LIMIT_DEFAULT 3
change this to
#define T316_ATTEMPT_LIMIT_DEFAULT 30
3. Increase maximum limit as according to default(set more than that of
default)
#define T316_ATTEMPT_LIMIT_MAX 10
change this to
#define T316_ATTEMPT_LIMIT_MAX 35
Thanks & Regards,
-
Akash Deep Verma
http://in.linkedin.com/in/akashdeep1
On Tue, Apr 16, 2013 at 4:13 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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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 at nms.co.in>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 at 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 at 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 at 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 at cupis.co.uk>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at simplicitymedialtd.co.uk>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 at lists.freeswitch.org [
>> freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael
>> Collins [msc at 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 at 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 at 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 at lists.freeswitch.org [
>>> freeswitch-users-bounces at lists.freeswitch.org] on behalf of Cal Leeming
>>> [Simplicity Media Ltd] [cal.leeming at 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 at 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 at 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:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> 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 at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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