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@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@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-calle...
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006232...
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-freeswi...
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.
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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 http://www.freeswitch.org
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