January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Prefix for outgoing calls
by Daniel Bareiro 09 Jan '11

09 Jan '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm starting to experiment with FreeSWITCH and I defined a gateway for outgoing calls to Iptel.org on the following file: /usr/local/freeswitch/conf/sip_profiles/external/iptel.org.xml using the following format: <include> <gateway name="iptel"> <param name="username" value="account"/> <param name="realm" value="iptel.org"/> <param name="password" value="password"/> <param name="proxy" value="sip.iptel.org"/> </gateway> </include> Apparently, FreeSwitch able to register against iptel: freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@10.1.0.45:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::iptel gateway sip:account@sip.iptel.org REGED external::ekiga gateway sip:account@ekiga.net REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@10.1.0.45:5060 RUNNING (0) 10.1.0.45 alias internal ALIASED ================================================================================================= 3 profiles 1 alias freeswitch@internal> Then I tried to make any call from the predefined extensions of FreeSwitch (1000.xml) to a number of iptel. Then calling from extension 1000 to, for example, music(a)iptel.org, I was able to communicate from a softphone. The reason why the call is successfully using @iptel.org is because it is taking the "realm" in the file iptel.org.xml? I have been reading the FS documentation but I not found a way to use a prefix to call a number of iptel (or some other provider) without having to use @provider. I have no clear the syntax I should use or where should I do this configuration. Someone could give me a hand with this configuration? Thanks in advance for your reply. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAk0qJgAACgkQZpa/GxTmHTeLsgCfWGwYeqV2iLixTSt/gaj6N/GC Q0AAnA/qTBANNifgvb6h6yhUgq2hxUEW =V4vj -----END PGP SIGNATURE-----

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portaudio no longer working
by covici＠ccs.covici.com 09 Jan '11

09 Jan '11

As of the latest git from today, port audio is no longer working. I get no sound and cannot even unload the module. When I shutdown, I have to kill fs with signal 9 and the last line in the log concerns pa. I am running linux gentoo x86. Any help would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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FreeSWITCH - Ploycom phone in hosted environment
by Aloysius Lloyd 09 Jan '11

09 Jan '11

Hi All, Ploycom phone's not supporting rport. How to use Ploycom phone in the Hosted environment. Polycom phone in a NAT environment and FreeSWITCH hosted in the Datacenter with public IP. I could not make it work. Any help is appreciated. Thanks Lloyd

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Use <tab> in mod_cdr_csv
by Boris Kovalenko 09 Jan '11

09 Jan '11

Hello! Is this possible to use special characters (<tab> for example) to separate fields? -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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Incoming call from registered gateway with DID in To: not URI
by Steven Ayre 08 Jan '11

08 Jan '11

Hi everyone, I have a gateway registering to numbergroup.com, this is the configuration: <gateway name="numbergroup"> <param name="realm" value="sip.numbergroup-services.com"/> <param name="username" value="XXXXXX"/> <param name="password" value="XXXXXX"/> <param name="register" value="true"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="60"/> <param name="ping-max" value="10"/> <param name="ping-min" value="1"/> <param name="retry-seconds" value="5"/> <param name="expire-seconds" value="60"/> </gateway> Incoming calls from my DID to a SIP URI on the server work fine. However I'm having problems sending it to the SIP Trunk (the registration above). The call arrives on the server fine, but the URI is the username(a)sip.numbergroup-services.com. The DID is in the To: header destination_number in the dialplan is the username not the DID as a result. Does anyone know how to configure the gateway so that the destination_number would contain the DID from the To header instead? Here's the INVITE: INVITE sip:username@81.27.101.246:5060;transport=udp;gw=numbergroup SIP/2.0 Via: SIP/2.0/UDP 80.93.165.111;rport;branch=z9hG4bKD4m88SNemDSBg Max-Forwards: 65 From: "+myphonenumber" <sip:+myphonenumber@80.93.165.111>;tag=y9DBpB33yBHNN To: <sip:+mydidnumber@sip.numbergroup-services.com> Call-ID: 9ee8ce2d-9544-122e-c2a7-002655d1d302 CSeq: 6881084 INVITE Contact: <sip:numbergroup@80.93.165.111:5060> User-Agent: numbergroup.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Session-Expires: 900 Min-SE: 90 Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 593 P-Charging-Vector: icid-value=f4571a9c-1aa1-11e0-8310-5798bfd3ad89;icid-generated-at=80.93.165… P-Asserted-Identity: "+myphonenumber" <sip:+myphonenumber@80.93.165.111> v=0 o=numbergroup 1294416066 1294416067 IN IP4 80.93.165.111 s=numbergroup c=IN IP4 80.93.165.111 t=0 0 m=audio 18230 RTP/AVP 8 9 98 3 18 99 100 101 102 103 104 0 105 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 SPEEX/8000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:102 G726-24/8000 a=rtpmap:103 G726-32/8000 a=rtpmap:104 G726-40/8000 a=rtpmap:105 CELT/48000 m=video 18540 RTP/AVP 106 107 34 31 a=rtpmap:106 THEORA/90000 a=rtpmap:107 H264/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 -Steve

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user variables
by Boris Kovalenko 08 Jan '11

08 Jan '11

Hello! I use FreeSWITCH Version 1.0.head (git-cd13030 2011-01-06 23-17-08 -0600) and have some troubles with user variables. So, when user is registered the user defined variables are present in channel variables, and when user isn't registered (authenticated by cidr for example) I can't see it variables. Is this my mistake or freeswitch bug? -- Regards, Boris

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server spec
by Rafqat . 08 Jan '11

08 Jan '11

Hi, I am interested in purchasing a server to host my calling card application (including freeswitch itself). The application is based on a simple lua script and which makes use of mod_lcr and mod_nibble. I would like to support at least 500 concurrent calls and was wondering what kind of server spec should I be looking to purchase. Any advice will be much appreciated. Cheers Raf

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Sip client with tunnel
by Budi wibowo 08 Jan '11

08 Jan '11

Hi I'm looking for sip client that support tunneling, because my isp seems block incoming sip connection. For sip outgoing no problem. Thx Budi

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cepstral
by Madovsky 07 Jan '11

07 Jan '11

Is anyone know if Cepstral is alive yet ? I bought voices and contacted support but no answer since 4 days now.

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hardware requirements
by ovvenkat 07 Jan '11

07 Jan '11

Hi to all, I have to handle 72,000 inbound calls per day ( 6 - 8 hours ), Anyone please suggest me the hardware requirement for the same. Like, What kind of Server and how many PRI or BRI cards will fulfill the needs. * Its Simple IVR, Call will be landing from mobile phones. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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