- FreeSWITCH-users - FreeSWITCH Mailing Lists

Multiple Registrations not working
by Yihui Li 14 Mar '11

14 Mar '11

Hi, I am trying to use Multiple Registrations feature in bridge, but found it not always working when there are more than 1 bridge targets like below: <action application="bridge" data="user/1015,user/1016" /> where 1015 is registered on 2 PC. To simplify my case, I have also tested it with CLI command. Test 1, originate user/1015 &parks both PC are ringing, Test 2, originate user/1015,user/1016 &parks only 1 PC is ringing in internal.xml, I added <param name="multiple-registrations" value="true"/> I am running git-head on ubuntu. Can anyone help with this issue? Thanks. Eric

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Can you use ext-ip.. autonat: with -nonat ?
by Mario G 14 Mar '11

14 Mar '11

Changed router now only 1 out of 5 calls come in. I previously had "ext-sip/rtp-ip:1.2.3.4" and -nonat on another router but having major issues on new one. Wiki does not answer this question: Can you use "exp-rtp/sip-ip"autonat:1.2.3.4" when freeswitch is started with -nonat ? Outbound is fine, inbound fails 80 percent if the time. I really wanted use ALG but when I did all worked fine except call dropped while conversing.

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multiple gateways and NAT
by Johan Helsingius 14 Mar '11

14 Mar '11

I am sure this question comes up all the time... I want to use several different SIP gateway providers (for different regions and services). Freeswitch seems to deal with NAT just great with one external gateway, but multiple gateways seems to be a rather thornier issue. So, do I need totally separate profiles for each gateway? And do they have to use different ports? And does that mean that I need to get my firewall to port map each of them separately (I would assume they all try to use the same 5060 port to reach my system in case of an incoming call)? Julf

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Problem with mapping new DIDs
by Prime 14 Mar '11

14 Mar '11

Hey! This most likely is a newb mistake but I'm stuck at the moment. Recently I had to add another provider to FS to handle incoming calls. And now I have a problem with calling my new provider's DID number on my FS. I looked at FS logs. Usually FS log says something like this " [INFO] mod_dialplan_xml.c:418 Processing caller_number->My_DID in context public" and then routes DID number to local extension, but now I see something like this " [INFO] mod_dialplan_xml.c:418 Processing caller_number->caller_number in context public ". Why new numbers does not match the regex? The number are configured exactly as old ones. Thanks in advance Best regards

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Socket stops receiving INVITE packets
by Steven Ayre 14 Mar '11

14 Mar '11

Hi everyone, I've got a strange problem on my servers which has happened 3 times in the last month now... I have FS listening on 2 IP addresses, bound to bond0 and bond0:0. FS suddently stops processing any calls arriving on one of those IPs. tcpdump/tshark still shows INVITE packets arriving on my server, but FS doesn't process them and there are no 100 Trying etc replies. Initially FreeSWITCH still owns the socket, the only odd thing is Send-Q is 744: ss1:~# netstat -anp | head -n2 Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name ss1:~# netstat -anp | grep 5060 tcp 0 0 81.27.101.252:5060 0.0.0.0:* LISTEN 27587/freeswitch tcp 0 0 81.27.101.233:5060 0.0.0.0:* LISTEN 27587/freeswitch udp 0 744 81.27.101.252:5060 0.0.0.0:* 27587/freeswitch udp 0 0 81.27.101.233:5060 0.0.0.0:* 27587/freeswitch I stop freeswitch so I can restart it: ss1:~# killall -9 freeswitch Freeswitch is no longer running (no zombies etc): ss1:~# ps faux | grep freeswitch root 20723 0.0 0.0 5164 788 pts/3 S+ 19:56 0:00 \_ grep --exclude-dir=.svn --color freeswitch But the port is still open! ss1:~# netstat -anp | grep :5060 udp 0 744 81.27.101.252:5060 0.0.0.0:* - This means I can't restart FreeSWITCH because the port is already in use! I have to reboot to free the port up. Has anyone seen this before? Does anyone know what could cause this? Since it's still in use after FS has stopped, I'm thinking a Linux kernel problem. I'm running Debian Lenny with kernel 2.6.26-2-amd64. The hardware is a NEC server with 2x E5405 xeon (8cores total) and 8gb ram. It's happened on 2 servers, so I'm thinking something in software. I updated kernel a couple of months back so perhaps that's related, before that everything was fine for 2 years. Thanks everyone... -Steve

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api_hangup_hook documentation correction required?
by Alex Crow 14 Mar '11

14 Mar '11

Anthony/all, I found while getting fax working that it seems that if api_hangup_hook is set /before/ answer in the following <extension name="Fax" > <condition field="destination_number" expression="^620$"> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="api_hangup_hook=system /usr/bin/php /var/www/fusionpbx/secure/fax_to_email.php email=acrow(a)integrafin.co.uk extension=620 name=\\\${last_fax} messages="/> <action application="set" data="last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}"/> <action application="rxfax" data="/opt/freeswitch/storage/fax/620/inbox/${last_fax}.tif"/> <action application="hangup"/> </condition> </extension> That rxfax strangely fails - the calling fax machine never gets the tone to start sending and T4 is retried until rxfax gives up. If it's placed as above it works fine. Not sure if this is a bug or intended and the docs on the Wiki need changing. Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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Freeswitch / Zywall USG100 problems
by Mario G 14 Mar '11

14 Mar '11

Just wondering if anyone here can help. I replaced a Linksys/Cisco RV042 router with a Zywall USG100. It solved a lot but broke Freeswitch badly. Yes I ran traces, etc. Both routers are set to use a static wan and no natting, ext-ip-sip/rtp are the wan static address. Just checking of anyone knows of a solution for one of these: 1. If I turn off ALG I cannot make a call or receive calls, the traces shows time outs. No firewall issues, nothing in any log to show the issue, 2. If I do have ALG on I can make calls but the HOLD button no longer works. Phone (Linksys SPA962) says "Trying to Hold" then times out. 3. With ALG on (so I can make calls) when I call out the call drops in the middle of a conversation between 30 seconds and 5 minutes. Been battling this and hope someone has already solved one of these. Thanks! Mario

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Re: [Freeswitch-users] mod_callcenter features
by Rob Hutton 14 Mar '11

14 Mar '11

1. Try something like: reloadxml callcenter_config queue load|reload queue_name 2. I don't believe so. Hit Moc on the IRC channel. I am trying, also... -- Thanks, Rob On Friday 31 December 2010 00:32:46 Rajkumar K wrote: > Hi, > > 1. In mod_callcenter, there is no command like "callcenter_config queue add" > for adding queues dynamically. We need to configure it in > callcenter.conf.xml file and run "callcenter_config queue reload <queue > name>". So is there any other ways to add queues. > > 2. The mod_callcenter automatically bridges the call once it identifies the > available agent. But the requirement is to identify the available agent and > leave the bridge control to our dial plan. There we need to other operations > (like announcing caller name to agent) instead of bridging the call. > > regards > rajkumar k > > On Thu, Dec 30, 2010 at 6:12 PM, Rob Hutton <justlikeef(a)gmail.com> wrote: > > > Some of the API commands are on in the Wiki under the heading API: > > > > http://wiki.freeswitch.org/wiki/Mod_callcenter > > > > Everything starts with callcenter_config, so you can feel your way around > > the command line and everything is fairly self explanatory. > > > > What do you mean control the call flow? Can you give an example of what > > you are trying to accomplish? > > > > -- > > Thanks, > > Rob > > On Monday 27 December 2010 04:42:29 rajkumar wrote: > > > > > > Hi, > > > > > > I am developing an application with mod_callcenter. I need to know the > > > following about mod_callcenter. > > > > > > * Is it possible to add/update/delete the queue configurations > > dynamically > > > without using static xml configurations. > > > * How can I control the call flow (for playback message and recording) > > > before and after bridging the call. > > > > > > Thanks in advance > > > > > > regards > > > rajkumar k > > > > > >

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Dialplan tone_detect problem
by Alex Crow 14 Mar '11

14 Mar '11

All, I am trying to implement fax detection on FreeSwitch within a dialplan that routes inbound calls from a BRI (Sangoma), to an Asterisk (Trixbox) machine handling the extensions. I am only doing this until I can get British English sounds for FS ;-). The inbound route without any fax detection works fine, ie the caller hears ringing until the Trixbox extension is either answered or hits voicemail: <extension name="60xto35x" > <condition field="source" expression="freetdm"/> <condition field="context" expression="public"/> <condition field="destination_number" expression="60([0-5])"> <action application="bridge" data="sofia/external/35$1(a)192.168.24.15"/> </condition> </extension> The closest I have got to adding the fax detection is the following: <extension name="60xto35x_copy" > <condition field="source" expression="freetdm"/> <condition field="context" expression="public"/> <condition field="destination_number" expression="60([0-5])"> <action application="tone_detect" data="fax 1100 r +5000 transfer '620 XML default'"/> <action application="answer"/> <action application="playback" data="tone_stream://$${uk-ring};loops=3"/> <action application="bridge" data="sofia/external/35$1(a)192.168.24.15"/> </condition> </extension> I had to add the playback so the caller would hear ringing during the tone_detect timeout. However, as soon as the tone_stream finishes, and the call is bridged to the Trixbox, the caller hears silence instead of ringing. Is there any way to pass the ringing from the extension back to the caller coming in on the BRI circuit, or otherwise fake it until the remote extension is answered? Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by Michel Habib 14 Mar '11

14 Mar '11

Sorry for the delay and thanks Chris, Please find attached logs: http://www.mediafire.com/?8thg9ey7kd26882 sip client ip 192.168.1.202 freeswitch ip 192.168.1.35 mrcp connector/speech server ip 192.168.1.32 i have attached 3 logs - freeswitch log file, wireshark on freeswitch server, wireshark on mrcp connector/speech server. my script simply plays back audio from wave [basic freeswitch function], then plays back audio from TTS [which cannot be heard] Thank you, Michel. ---------- Forwarded message ---------- > From: Christopher Rienzo <cmrienzo(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 9 Mar 2011 22:25:34 -0500 > Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server > [using MRCP Connector] - Audio Problem > I still would like to see the logs for your call. > > > On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib(a)gmail.com>wrote: > >> Yes, I get the Audio from FS in regular calls - I already disabled all >> possible firewalls - all 3 machines [softphone, freeswitch, Speech Server >> (and mrcp connector) ] are on a switch. >> 192.168.5.107 is the freeswitch server >> 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server >> I made too many iterations on the configuration below: >> >> <include> >> <profile name="mrcp-connector" version="2"> >> <param name="client-ip" value="192.168.5.107"/> >> <param name="client-port" value="5090"/> >> <param name="server-ip" value="192.168.5.110"/> >> <param name="server-port" value="5070"/> >>  >> <param name="sip-transport" value="udp"/> >>  >>  >> <param name="rtp-ext-ip" value="192.168.5.107"/> >> <param name="rtp-ip" value="192.168.5.107"/> >> <param name="rtp-port-min" value="4000"/> >> <param name="rtp-port-max" value="5000"/> >>  >> <param name="rtcp" value="1"/> >>  >> <param name="rtcp-bye" value="2"/> >>  >> <param name="rtcp-tx-interval" value="5000"/> >>  >> <param name="rtcp-rx-resolution" value="1000"/> >> >>  >>  >>  >> <param name="codecs" value="PCMU PCMA L16/96/8000"/> >> >>  >> <synthparams> >> </synthparams> >> >>  >> <recogparams> >>  >> </recogparams> >> </profile> >> </include> >> >> >> >> ---------- Forwarded message ---------- >> >>> From: Christopher Rienzo <cmrienzo(a)gmail.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Date: Mon, 7 Mar 2011 09:32:51 -0500 >>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server >>> [using MRCP Connector] - Audio Problem >>> Do you get audio between FS and your SIP client when not using ASR/TTS? >>> >>> Show me the MRCP profile configuration and your FreeSWITCH logs during >>> the call. >>> >>> >>> >>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib(a)gmail.com>wrote: >>> >>>> Hello All, >>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP >>>> calls and use its ASR and TTS Services successfully] >>>> I am also using MRCP Connector from AumTech - which allows me to use ASR >>>> and TTS Services through an MRCP Client . >>>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS. >>>> >>>> for TTS, I can successfully make the call, the Audio RTP of the TTS >>>> voice is transferred succesfully from Speech Server [through MRCP Connector] >>>> back to the Freeswitch Server. >>>> However, Freeswitch is not sending back the Audio RTP to the SIP client. >>>> >>>> for ASR, I can successfully define the grammar and start recognition, >>>> but the audio RTP sent to speech server [through MRCP Connector] is silent >>>> [empty]. >>>> >>>> I am suspecting something is wrong with the RTP Configuration - can you >>>> help me? >>>> >>>> Let me now if you need any specific logs/scripts/configuration? >>>> >>>> Thank you, >>>> Michel. >>>> >>>

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