- FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound dialplan
by Baskar 28 May '11

28 May '11

Hi All,, In my inbound dial plan xml file i have set two conditions enabled Condition1: In inbound dial plan callers are given an option to key in the extension number and reach the appropriate extension (Example: 1001 or 1002 or 1003 etc). Condition2: The second condition routes call to a default extension in scenarios where the caller does not specify any extension number (Example: Default extension is 1007). Both the above conditions are working fine. Now I need to set up another condition where after keying in the extension number the call gets transferred to the appropriate extension and if the extension is busy(Example: say extension 1003 is busy) it should be hunted to a default extension (example: 1007). How can we set up this condition in dial plan? Can any one guide me. -- Thanks with Regards, N.Baskar

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Received in VIa
by Spencer Thomason 27 May '11

27 May '11

Hello all, We are trying to use Freeswitch as a SBC in conjunction with Opensips. The call flow looks like this User UA -> Opensips -> ITSP | | \/ /\ Freeswitch Opensips acts as a registrar and select the outbound route and listens on X.X.X.X:5060. Freeswitch is a B2BUA and signaling server with media and listens on X.X.X.X:5070. Basically a user sends an INVITE and immediately following auth, Opensips forwards to Freeswitch which sends a new call back to opensips which is then routed to our ITSPs. With FS not listening on the default port, we get a received in the Via: message. Via: SIP/2.0/UDP X.X.X.X:5070;received=X.X.X.X;rport=5070;branch=z9hG4bKpU7KvSaeKSc5D Is there any way to remove the received header? Thanks, Spencer

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matching word
by Sam 27 May '11

27 May '11

Friends, How can i match words in expressions, is it by expression="^(\y\w+\y)$" or could be something else. Regards Sam

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mod_fsv, record and playback options
by William Kendi ... 27 May '11

27 May '11

Greetings, Currently, I am trying to record and play videos through FreeSWITCH using the "mod_fsv" module. Some options like timeouts and interrupts by dtmf digits seems to be lacking in the "mod_fsv" module and I cant even stop the video recording using the "break" application. The video recording stop when the user disconnects. Anyone has any ideas about how to stop the FSV video recording without disconnecting the user? Thanks in advance.

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event inconsistency
by Adam Kelloway 27 May '11

27 May '11

Hi there, Has anyone ever had problems with events not being sent out over socket connections reliably? I am noticing that sometimes event notifications are sent, and sometimes they are not. For example, I have subscribed to CHANNEL_HANGUP, but depending on the timing of when I hangup the call (or when the dial plan hangs up), the event is not received. I confirmed with tcpdump that it is not even sent over the socket. Has anyone else seen his? What might be the problem? Thanks

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Re: [Freeswitch-users] FreeSWITCH as SIP softswitch
by Cyril Zlachevsky 27 May '11

27 May '11

Can't understand you Answer, Ivan! You mean I have to replace <action application="bridge" data="sofia/gateway/Asotel/$1"/> to <action application="bridge" data="sofia/directory/Asotel/$1"/> in conf/dialplan/public/test.xml file? 2011/5/27 <freeswitch-users-request(a)lists.freeswitch.org>: > From: "Иван Чистяков" <zetruger(a)gmail.com> > Use directory for SIP clients (hardphone, softphone) instead gateways. > Use gateway just for SIP providers (SIP trunks). > > Simple test (run from FS console): > originate sofia/internal/outboundtest@x.x.x.x:p > &bridge(sofia/internal/inboundtest@x.x.x.x:p) > > 2011/5/27 Cyril Zlachevsky <cyril.zlachevsky(a)gmail.com>: >> Hello, >> I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server. >> But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.

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Codec selection for incoming call
by Łukasz Zwierko 27 May '11

27 May '11

Hi guys, Is is possible to manually select which codec will be used on the A-leg (incoming call) when using outbound event socket to control the call? My case is quite basic I guess: I want to answer the call and pass it to an application which will record raw data (so I'm not bridging the call anywhere). It is important for me to be able to make a per-call selection of codec. I've been digging through the wiki the whole morning, but I've only found out how to force a codec on the outgoing call... thanks for any advice, Lukasz

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Virtual Migration Detected! - Crit errors in CLI
by clive engelberg 27 May '11

27 May '11

Hi I hope someone can help me with this issue. It seems that there are clock slippages due to overworked CPU with heavy load. I am getting these errors with 2 or 3 channels only. On closer inspection, it seems that the server (dell 1950) is using only 1 CPU. The interrupts show that the other CPU's are hardly being used. (see the cat/proc/interrupts below). my gut feeling is that the APIC is not working as it should. Would changing irq's in the bios help this issue? I am running Centos 5.5 Any suggestions will be appreciated. Thanks in advance Clive # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 0: 782410882 0 0 0 0 0 0 0 IO- APIC-edge timer 1: 2 0 0 0 0 0 0 0 IO-APIC- edge i8042 8: 1 0 0 0 0 0 0 0 IO-APIC- edge rtc 9: 0 0 0 0 0 0 0 0 IO-APIC- level acpi 12: 4 0 0 0 0 0 0 0 IO- APIC-edge i8042 14: 238 194 0 0 7005401 15210 0 0 IO-APIC-edge ide0 66: 23 0 0 0 0 0 0 0 IO- APIC-level ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb4 74: 0 0 0 0 0 0 0 0 IO-APIC- level uhci_hcd:usb3 82: 5426 137633 0 147604 1635602 0 0 0 IO-APIC-level megasas 90: 6118176 0 0 0 0 0 0 0 PCI-MSI eth0 98: 3457897 0 0 0 0 0 0 0 PCI-MSI eth1 NMI: 6258 2735 3482 3453 3253 2513 3379 2652 LOC: 782410253 782410193 782410118 782410043 782409968 782409893 782409818 782409736 ERR: 0 MIS: 0 ____________________________________________________________ South Africas premier free email service - www.webmail.co.za Save on insurance with OUTsurance https://www.outsurance.co.za/insurance-quote/?source=webmailmailer&cr=genp1…

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FreeSWITCH as SIP softswitch
by Cyril Zlachevsky 27 May '11

27 May '11

Hello, I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server. But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help. FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite softphone with dynamic IP from subnet 92.112.0.0/16) and forward this calls to the hardware SIP phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer. Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial 7777. But through FreeSWITCH calls always fail. I created four configuration files - all other configs I left unchanged: 1) conf/sip-profiles/internal/X-Lite.xml: <include> <gateway name="X-Lite"> <param name="username" value="inboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include> 2) conf/sip-profiles/external/Asotel.xml: <include> <gateway name="Asotel"> <param name="realm" value="195.225.XXX.XXX"/> <param name="username" value="outboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include> 3) conf/dialplan/public/test.xml: <include> <extension name="test"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="sofia/gateway/Asotel/$1"/> </condition> </extension> </include> 4) conf/directory/default/inboundtest.xml: <include> <user id="inboundtest" cidr="92.112.0.0/16"> <params> <param name="from-domain" value="88.198.XXX.XXX"/> <param name="password" value="test"/> </params> </user> </include> This debug from Asotel ip-phone: ---begin--- Incoming CallLeg at callleg created 0x57d334 Incoming CallLeg at MsgReceived 0x57d334 *** $1 was being Invited *** >>> All call occupied. <<< No slot availabe for this call... FindIPCall...All Slot is Busy RvSipCallLegReject(486), hCallLeg: 57d334 --> Message Sent (Message type: 1) (call-leg 57d334) SIP/2.0 486 Busy Here From: "inboundtest"<sip:outboundtest@195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p To: <sip:$1@195.225.XXX.XXX>;tag=c3e19fb6-13c4-4ddecb5a-1f577-5f2a Call-ID: 232b19fe-0285-122f-b5b5-1b5bdf4f9807 CSeq: 12887277 INVITE Via: SIP/2.0/UDP 88.198.XXX.XXX:5080;rport=5080;branch=z9hG4bKv0XUcNtvK3c2K Supported: replaces User-Agent: FXS_GW (1asipfxs.109) Content-Length: 0 ---end--- In freeswitch log I can see this: ---begin--- [NOTICE] switch_channel.c:816 New Channel sofia/internal inboundtest(a)88.198.XXX.XXX [bf8e8081-eaf1-453e-a643-ee03df36ba0f] [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->7777 in context public [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [4beaba1f-c9c6-4ed7-94c5-efec453e895a] [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [USER_BUSY] [INFO] mod_dptools.c:2685 Originate Failed. Cause: USER_BUSY [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest(a)88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest(a)88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest(a)88.198.XXX.XXX [CS_DESTROY] [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY] ---end--- In my last tests I called to the voip-provider test number and got UNALLOCATED_NUMBER disconnect cause: ---begin--- [NOTICE] switch_channel.c:816 New Channel sofia/internal/inboundtest(a)88.198.XXX.XXX [ee9e4e33-0676-4c9c-9952-ff97c4d8db18] [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->555 in context public [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [b8278ac0-8e90-4d48-bdbf-8d0f608ec35a] [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] [INFO] mod_dptools.c:2685 Originate Failed. Cause: UNALLOCATED_NUMBER [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest(a)88.198.XXX.XXX [CS_EXECUTE] [UNALLOCATED_NUMBER] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest(a)88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest(a)88.198.XXX.XXX [CS_DESTROY] [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY] ---end--- What am I missing here? Thanks for your help.

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Errors while sending faxes
by Mathieu Lautram 27 May '11

27 May '11

Hi all, After sending faxes (it works very well know, thanks to Anthony Minessale) , sometimes I've got errors like that: - Disconnected after permitted retries - No response after sending a page - Received a DCN from remote after sending a page - Received bad response to DCS ot TCF - The call dropped prematurely - Timed out waiting for initial communication Those faxes are about 25% of all faxes that I send. I'm using the last git version. Where this problem comes from? Is there a way to fix this? I know that sometimes the remote end could be broken or could be an old fax machine, but it seems strange that I have so many errors like that. Thank you for your help :-) -- Mathieu LAUTRAM Application developer BJT Partners - FRANCE +33 1 79 75 99 60 +33 6 61 59 07 25

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