July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Calling an extension on a remote freeswitch box
by Yungwei Chen 06 Jul '11

06 Jul '11

Hi, I'm trying to place a call to an extension on a remote freeswitch box. I tried the following command, but it doesn't work. originate {ignore_early_media=true}sofia/internal/77777(a)192.168.1.20 test Here's the error message. 2011-07-05 18:06:46.670942 [ERR] sofia_reg.c:1912 Cannot locate any authentication credentials to complete an authentication request for realm '"192.168.1.18"' Am I using the right dial string? If so, how can I fix the error? Thanks.

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GTalk mod_dingaling read and send text messages
by Sebastien Peterson 05 Jul '11

05 Jul '11

Hi freeswitch-users, I am a new FreeSwitch user. I am trying to send and read Gtalk text messages from the dialplan. Is that something I can do and how? Could not find examples from the wiki. Once I read the message, I need to place it into a variable. In advance thanks ! Seb

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NAT Traversal
by Carmelo Fischetti - Digi Unit 05 Jul '11

05 Jul '11

Hi all, I'm experiencing a NAT traversal problem connecting some softphones to a FreeSWITCH server. At office the softphone registers with TLS-NAT, while at home it registers with TLS and actually it's unreacheable (it can place calls but it cannot receive them), even if the registration parameters (sofia status profile internal reg) contain both the public (IP: x.y.z.w) and the private (Contact: "user" <sip:1010@192.168.1.100:6010;transport=TLS;line=3c4e8d673906c5d>) IP addresses. In both cases the softphone is behind a router, the server is behind a firewall and runs the ddclient service (there is a DynDNS name assigned to it). Am I missing some configuration detail at server side? At the moment I've modified the default configuration files in the following way. The following lines in vars.xml: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> were substituted with: <X-PRE-PROCESS cmd="set" data="domain=myhost.dyndns-server.com"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:myhost.dyndns-server.com"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:myhost.dyndns-server.com"/> The following lines in sip_profiles/internal.xml: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> were substituted with: <param name="ext-rtp-ip" value="host:myhost.dyndns-server.com"/> <param name="ext-sip-ip" value="host:myhost.dyndns-server.com"/> and I've also added the line: <param name="sip-force-contact" value="NDLB-connectile-dysfunction"/> but I actually don't understand what it means (I've found it in some posts about NAT traversal in FreeSWITCH). Thanks in advance. Carmelo

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pl HELP! how to change conference's caller-id-number at run-time?
by ran zhang 05 Jul '11

05 Jul '11

Hi All: I want when the bridging conference is inviting someone to start the conference, the caller is shown as the real caller id number rather than whats stored in 'caller-id-number' in conference.conf.xml. I have tried to change the 'effective_caller_id_number', it only affects when its a bridging call.

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originate a call to an extension and then connect to another extension
by Yungwei Chen 05 Jul '11

05 Jul '11

Hi, I am wondering if it's possible to originate a call to an extension and then connect to another extension. If so, how does the the dialstring look like? Thanks.

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Outbound socket, bridge command - how to specify leg_timeout?
by Mateusz Bartczak 05 Jul '11

05 Jul '11

Hi all I use outbound socket to handle call, I'm bridging call to external voip provider. This works very nice but I also need to specify call timeout (once answered). I've tried following but with no success: sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true,leg_timeout=10}sofia/gateway/trunk/600 event-lock: true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true}[leg_timeout=10]sofia/gateway/trunk/600 event-lock: true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: [leg_timeout=10]sofia/gateway/trunk/600 event-lock: true Leg_timeout seems to be ignored and call continues after 10 seconds. What's the correct form of setting it for calls bridged via outbound socket handler? Is the leg_timeout a good choice to limit maximum answered call length?

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Skeleton Configuration?
by Bryan Smart 05 Jul '11

05 Jul '11

Is a truly skeletal configuration available? As far as I can tell, the choice right now is to start with no configuration files at all, or else to install a sample set that is bursting with stuff. So far, I've installed the samples, and tried to strip out the parts that are obviously not needed by me: The dialplan is the best example that I can think of: all the exts that replace functions on Snom/Linksys phones, record/play video, demo substitute ring tone, call groups, add/del groups, dynamically create numerous types of conferences, call features like redial/hold/etc, the exts for the demo IVR, and so on. I know enough to know that I don't want those on a live system, but I don't know enough to know if it is safe to remove parts. Will I break refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary as a safety feature if I'm connecting sessions to SIP URIs outside of the local FS? ? Can I drop "external-sip-uri" if users won't be calling external URIs, or will that completely block FS from connecting to external URIs in all cases? I think I can dump features.xml altogether. There are an internal and external SIP profile. If I'm only hosting a voice application, I think I can dump the internal profile and just run the default external profile on 5060, but there are a huge amount of options in there that look as if they're setup to cover cases where the external profile is used to connect to ITSPs, which I won't. There are so many configuration files that I'm not comfortable with blowing the whole directory tree away and recreating a raw basics version of each of them. While I don't feel paranoid about a specific part of the security of the full samples install, my rule is always that I try to not run anything that I don't at least somewhat understand. There seem to be so many little demos and optional features that I know that I by no means understand the implications of what they could be used to do in combination with what I'll add later. It seems safer to start from the bare basics, and only add what is needed, but, for now, I must start with the samples, and try to strip away as much as I can without breaking FS. For me, I'm using FS as a platform for running a voice app, so I need little, if any, PBX type features. Some people that might wish to use FS as a PBX might not want all of the demo bits, though. An absolute minimum set of configuration files, as well as a set containing the essentials for a app platform or PBX would be useful, if they don't already exist somewhere. Bryan

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Video is not display
by sc_zhangming 05 Jul '11

05 Jul '11

hi: if use fs_cli originate user/70001 700002 agent is not vedio if i use x-lite 4 input number the call have vedio my dialplan configuration information: <condition field="destination_number" expression="^(7[0-4][0-9][0-9][0-9][0-9])$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/$1@${domain_name}"/> <action application="answer"/> </condition> </extension> log: 2011-07-07 19:04:31.427458 [DEBUG] sofia.c:4669 Remote SDP: v=0 o=- 12954249923203125 3 IN IP4 10.108.226.238 s=CounterPath X-Lite 4.0 c=IN IP4 10.108.226.238 t=0 0 a=ice-ufrag:2e4e41 a=ice-pwd:1af7090a62cdba6f63b4b6bd9fd01472 m=audio 64370 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=candidate:1 1 UDP 659136 10.108.226.238 64370 typ host a=candidate:1 2 UDP 659134 10.108.226.238 64371 typ host m=video 63720 RTP/AVP 34 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=2 a=candidate:1 1 UDP 659136 10.108.226.238 63720 typ host a=candidate:1 2 UDP 659134 10.108.226.238 63721 typ host 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:2690 Already using PCMU 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send payload to 101 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4626 Video Codec Compare [H263:34]/[H263:34] 2011-07-07 19:04:31.427458 [DEBUG] sofia.c:5121 Processing updated SDP 2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2535 (sofia/internal/sip:700001@10.108.226.222:39206) Callstate Change ACTIVE -> HANGUP 2011-07-07 19:04:36.870143 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:700001@10.108.226.222:39206 [CS_HIBERNATE] [NORMAL_CLEARING] 2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/sip:700001@10.108.226.222:39206 [KILL] sc_zhangming(a)sina.com.cn

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MSVC 2010 Compile broken
by Jan Berger 03 Jul '11

03 Jul '11

Hi, I just downloaded the latest FreeSWITCH and I have not been able to compile it with Visual Studio 2010 so far. Firstly - opening FreeSWITCH.2010.sln gives an error - a missing project. Ignoring that and switching off spandsp I still end up with compiler errors in Sofia. Opening and running FreeSWITCH.2010.express.sln or converting FreeSWITCH.2008.sln gives the same result - Sofia is the first that don't compile. The actual error I get is that http_host_class is undefined and checking with the original Sofia I notice that http_protos.h that should have contained this is empty ??? I assume it's a build step that don't work or have dropped out - any ideas? Jan

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Invitation to connect on LinkedIn
by Jacob Maldonado via LinkedIn 03 Jul '11

03 Jul '11

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