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Re: [Freeswitch-users] IVR, bridge, ring back problem
by andpe 16 Apr '13

16 Apr '13

Dziękuję bardzo, wszystko działa tak jak opisałeś. Pozdrawiam W dniu 2013-04-16 07:29:46 użytkownik Michael Collins <msc(a)freeswitch.org> napisał: I believe it's the space and not the colon. However, I don't personally recommend that you put everything into the IVR like that. Instead, try putting all those set apps and the bridge app into a separate XML dialplan extension and then use the transfer app to send the caller there. See the demo IVR in conf/ivr_menus/ for examples on how to do that. -MC On Fri, Apr 12, 2013 at 10:12 PM, Andrzej <andpe(a)poczta.onet.pl> wrote: I'll check on Monday. but: 1. Is this normal behavior IVR? 2. How it should be written: a) <entry action="menu-exec-app" digits="1" param="set ringback=$${uk-ring}"/> <entry action="menu-exec-app" digits="1" param="set instant_ringback=true"/> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/xxx"/> or b) <entry action="menu-exec-app" digits="1" param="set:ringback=$${uk-ring}"/> <entry action="menu-exec-app" digits="1" param="set:instant_ringback=true"/> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/xxx"/> look at the colon after "set". Regards, W dniu 12.04.2013 18:25, Paul Cupis pisze: > On Fri, Apr 12, 2013 at 02:07:45PM +0200, andpe wrote: >> I have a problem with the IVR in FreeSwitch and function bridge. >> >> How it works: >> customer calls IVR, IVR announses >> customer chooses button 1 >> number 1 uses function bridge to connect to, say, 1001. >> during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). >> when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. >> I have seen in logs Ring Ready not send to calling party. >> >> When user calls directly 1001 all works fine. > Do something like this before the bridge? > > <entry action="menu-exec-app" digits="1" param="set ringback=$${uk-ring}"/> > <entry action="menu-exec-app" digits="1" param="set instant_ringback=true"/> > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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(a)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

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 82, Issue 133
by Akash Deep Verma 16 Apr '13

16 Apr '13

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(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > 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(a)nms.co.in> > To: FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)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(a)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(a)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(a)cupis.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)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(a)simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)lists.freeswitch.org [ >> freeswitch-users-bounces(a)lists.freeswitch.org] on behalf of Michael >> Collins [msc(a)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(a)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(a)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(a)lists.freeswitch.org [ >>> freeswitch-users-bounces(a)lists.freeswitch.org] on behalf of Cal Leeming >>> [Simplicity Media Ltd] [cal.leeming(a)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-call… >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/00623… >>> >>> 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-freesw… >>> >>> Hope this helps >>> >>> Cal >>> >>> On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt < >>> jeff(a)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(a)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(a)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(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)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(a)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(a)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(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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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ringback and transfer_ringback ignored by FS
by Adam Raszynski 16 Apr '13

16 Apr '13

Hi Scenario: - SIP only - inbound call, hits dialplan - call answered, some IVR used - call bridged to selected department The problem is, that I would like to play music on hold while trying to bridge. FS seems to completely ignore custom ringback settings: <action application="set" data="ringback=myfile.wav"/> <action application="set" data="transfer_ringback=myfile.wav"/> <action application="bridge" data="{ignore_early_media=true}sofia/internal/x@y" /> tried also with: <action application="bridge" data="{ignore_early_media=true,transfer_ringback=myfile.wav,ringback=myfile.wav)}sofia/internal/x@y" /> and <action application="set" data="ringback=myfile.wav"/> <action application="set" data="transfer_ringback=myfile.wav"/> <action application="ring_ready" /> <action application="bridge" data="{ignore_early_media=true}sofia/internal/x@y" /> All with no success. I see in console that variables are set, but for some reason FS ignores them and there is no audio while bridging. At the same time I can play/stream sound file before bridge, but this operation is blocking and stream can not be run in background so it can't be used for MoH purpose. Any hints what should I check or how to achive nonblocking MoH?

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Re: [Freeswitch-users] PRI channels suspended
by Ashish gautam 16 Apr '13

16 Apr '13

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(a)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(a)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(a)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(a)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 >>> >>> >> >

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PRI channels suspended
by Akash Deep Verma 16 Apr '13

16 Apr '13

Hi Ashish, This is a hack but you 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

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Newbie question about inbound calls/NAT (yes I checked wiki first:-) )
by Jeff Bernhardt 16 Apr '13

16 Apr '13

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(a)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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IVR, bridge, ring back problem
by andpe 16 Apr '13

16 Apr '13

Welcome. I have a problem with the IVR in FreeSwitch and function bridge. How it works: customer calls IVR, IVR announses customer chooses button 1 number 1 uses function bridge to connect to, say, 1001. during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. I have seen in logs Ring Ready not send to calling party. When user calls directly 1001 all works fine. Another question: How to pass music to user when hi waiting for connection when I use bridge function in IVR? I was looking for a solution but I failed to find any. I am new in FreeSwitch. Sory for my english. Regards Andy

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FS Wiki site is down
by Muhammad Shahzad 16 Apr '13

16 Apr '13

It seems FS Wiki site is down at the moment. I am getting this on error page, Can't contact the database server: Unknown MySQL server host ' db.freeswitch.org' (2) (db.freeswitch.org) Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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how Boost fs service startup
by shayne.alone＠gmail.com 16 Apr '13

16 Apr '13

Hi, how can I find the reason cause freeswitch startup service, to talk long too mutch? and how can I help it's startup process to go faster? thanks

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PAP2T INCOMPATIBLE_DESTINATION
by Juan Antonio Ibañez Santorum 16 Apr '13

16 Apr '13

Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? Regards

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