April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nuance mod_unimrcp ASR
by Anita Hall 16 Apr '12

16 Apr '12

Hi I am making a FreeSWITCH application for ASR using Nuance Speech Server 5.0 and mod_unimrcp. What are my options ? regards, Anita

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Help with tone detection
by Jean-Marc Hyppolite 16 Apr '12

16 Apr '12

Hello everyone, Is it possible to have some help in setting up mod_spandsp for call progress tone detection? I have not been successful thus far. I keep getting an "error reading frame" message like this one below. ============================================================================ =================== 2012-04-09 16:03:22.550303 [DEBUG] mod_spandsp_dsp.c:379 (sofia/internal/sip:1000@184.162.247.104:32986) Starting tone detection for '1' 2012-04-09 16:03:22.550303 [INFO] mod_spandsp_dsp.c:411 (sofia/internal/sip:1000@184.162.247.104:32986) initializing tone detector 2012-04-09 16:03:22.550303 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/internal/sip:1000@184.162.247.104:32986 2012-04-09 16:03:22.590301 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:422 (sofia/internal/sip:1000@184.162.247.104:32986) error reading frame 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:447 (sofia/internal/sip:1000@184.162.247.104:32986) destroying tone detector ============================================================================ =================== My spandsp.conf.xml file: ============================================================================ =================== <configuration name="spandsp.conf" description="Tone detector descriptors"> <descriptors>    <descriptor name="1"> <tone name="CED_TONE"> <element freq1="2100" freq2="0" min="500" max="0"/> </tone> <tone name="SIT"> <element freq1="950" freq2="0" min="256" max="400"/> <element freq1="1400" freq2="0" min="256" max="400"/> <element freq1="1800" freq2="0" min="256" max="400"/> </tone> <tone name="REORDER_TONE"> <element freq1="480" freq2="620" min="224" max="272"/> <element freq1="0" freq2="0" min="224" max="272"/> </tone> <tone name="BUSY_TONE"> <element freq1="480" freq2="620" min="464" max="516"/> <element freq1="0" freq2="0" min="464" max="516"/> </tone> </descriptor> </descriptors> </configuration> ============================================================================ =================== Thank you for your help. Jean-Marc.

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external sip profile - registered users get Originate Failed. Cause: USER_NOT_REGISTERED
by Karsten Horsmann 16 Apr '12

16 Apr '12

Hi all, i try to install FS as SBC between public internet and private subnet. (SIP-User)--->[public-ip]FreeSWITCH[private-ip]--->(IVR System) If i try to reach an user, i got USER_NOT_REGISTERED. If i registered to the internal sip profile with the private address, everything works fine. Any hints? freeswitch@kamailio1> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 212.xx.xx.xx SIP-IP 212.xx.xx.xx URL sip:mod_sofia@212.xx.xx.xx:5060 BIND-URL sip:mod_sofia@212.xx.xx.xx:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 4 FAILED-CALLS-IN 2 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 2 freeswitch@kamailio1> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 212.59.42.189,172.20.100.74 Dialplan XML Context public Challenge Realm auto_from RTP-IP 172.20.100.74 SIP-IP 172.20.100.74 URL sip:mod_sofia@172.20.100.74:5060 BIND-URL sip:mod_sofia@172.20.100.74:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 1 FAILED-CALLS-OUT 1 REGISTRATIONS 0 -- Mit freundlichen Grüßen *Karsten Horsmann*

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Integrating FreeWITCH and Opensips
by azza miled 16 Apr '12

16 Apr '12

Hi! I want to use FreeSWITCH to handle IVR for my opensips's users.I have no idea where to start.A detailed tutoriel will be helpful. Can any one help me, please?

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FreeSWITCH Community Conference Call: DNS Best Practices in a VoIP Environment
by Michael Collins 16 Apr '12

16 Apr '12

Greetings, We'd like to have everyone join us on Wednesday, April 18th for a great presentation to be given by our very own Travis Cross. He will be discussing DNS Best Practices in a VoIP Environment. I'm definitely looking forward to this discussion as this is an important topic and formal documentation isn't too widely dispensed at the moment. I will be taking notes on this discussion and creating a corresponding wiki page. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_18 To make this discussion more meaningful I would like everyone to be willing to share their personal experiences and knowledge, as well as any questions that they have on the subject. We have a number of community members whose skill we can draw upon, and the more questions we ask, the more information we can collect and put into formal documentation. Talk to you this Wednesday! -Michael

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Faxing via FreeSWITCH documentation
by Timothy Bolton 16 Apr '12

16 Apr '12

Hey all, I know that there is a good amount of information out there on how to do faxing via FreeSWITCH, but I'm still finding myself a bit confused. I'm currently trying to figure out the process by which I would need to do this. I'll detail my setup, and ask questions. I've read documentation, but I'm still missing something. Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM, and I'm the only user. I'm using FlowRoute, and I have a couple of unused numbers. Pretty simple, nothing out of the ordinary (except maybe the VM part?). At some point in the next six months or so, I'll be getting a couple of dedicated servers here for various things, and I'll begin moving my VM servers from my main computer to those, as actual hardware hosts. So my fax machine is built into an "All-in-one" Network Printer/Scanner/Fax (Brother MFC-7360N). Here's my main question, do I need to get a special card to run a cable from my computer to the Fax? Is it better to get a dedicated fax machine to use, or are there specialized fax machines for Fax over IP? Thanks guys and gals, Tim -- 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed

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mod_callcenter - using moh and ring tones when phone is ringing; possible?
by Fahrwerk Kurierkollektiv (Technik) 16 Apr '12

16 Apr '12

Hi all, Is it possible to use moh and if a phone is ringing to play ring tones to the calling party? Thanks in advance, Georg

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Authenticate external calls problem
by Jose Miguel Sucasas Mejuto 16 Apr '12

16 Apr '12

Hello, I have installed freeSwitch 1.1beta from the git repository (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000). My SIP provider (Netelip) is defined in conf/directory/default/example.conf.xml: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/> <param name="auth-calls" value="false"/>  <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="$${default_provider_password}"/> </params> </user> </include> and doesn't authenticate inbound calls (insecure=invite in asterisk language). When I receive an inbound call, I receive the following error: freeswitch@internal> recv 1000 bytes from udp/[194.140.135.80]:5060 at 23:23:10.163377: ------------------------------------------------------------------------ INVITE sip:34810101189@sukmad.dyndns-home.com SIP/2.0 Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com> Contact: <sip:916495242@194.140.135.80> Call-ID: 6c47a4064ef8d0641da415d2121e3226(a)194.140.135.80 CSeq: 102 INVITE User-Agent: Netelip Max-Forwards: 70 Date: Fri, 13 Apr 2012 23:23:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 425 v=0 o=root 1249 1249 IN IP4 194.140.135.80 s=session c=IN IP4 194.140.135.80 t=0 0 m=audio 16474 RTP/AVP 8 0 97 3 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 382 bytes to udp/[194.140.135.80]:5060 at 23:23:10.164409: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060 From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com> Call-ID: 6c47a4064ef8d0641da415d2121e3226(a)194.140.135.80 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000 Content-Length: 0 ------------------------------------------------------------------------ 2012-04-14 01:23:10.162331 [DEBUG] sofia.c:7598 IP 194.140.135.80 Rejected by acl "domains". Falling back to Digest auth. send 871 bytes to udp/[194.140.135.80]:5060 at 23:23:10.166026: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060 From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com>;tag=SaBHcvmm7evNF Call-ID: 6c47a4064ef8d0641da415d2121e3226(a)194.140.135.80 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="194.140.135.80", nonce="bc1571d4-0957-4609-a887-7c3359b9ae9c", algorithm=MD5, qop="auth" Content-Length: 0 I have read in forums that the solution to this problem is to add the IP to the acl.conf.xml, like this: <list name="domains" default="deny">  <node type="allow" domain="$${domain}"/>  <node type="allow" cidr="194.140.135.80/24"/> </list> But my SIP provider have a dynamic IP, and it's behind a NAT. If my SIP provider will use any IP, ¿Where can I do this without enabling all IPs to enter calls in my system? Thanks in advance, Regards.

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Bridged call not ending/terminating
by ocset 15 Apr '12

15 Apr '12

Hi I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway. I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up. Here is what I have tested 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up. Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation? I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active. ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way. Thanks in advance O.

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Fwd:
by Manjiri Deshpande 15 Apr '12

15 Apr '12

http://cityspots.ro/play.php?jvuwys=481&anjgenax=11

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