September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Case insensitive SIP URI
by Serge S. Yuriev 20 Sep '12

20 Sep '12

Hello, We have multiple registrations turned on on both profiles and few of our users registered at Internal, some on External. Via sofia_conntact we are able to reach them all BUT some registers for example as Support(a)domain.tld and others as support(a)domain.tld - and this is big problem for us :( I believe it's incorrect behavior to match case sensitive although it's "by design" (as per Jira). Anyone can propose workaround or patch for Sofia? Thanx in advance. -- wbr, Serge

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The subscriber appears to absent, but only sometimes and at some locations.
by Gabriel Gunderson 20 Sep '12

20 Sep '12

All, I have a setup (happens to be domain base multi-tenancy) where we occasionally have an issue where 1 or 2 (sometimes more) of the extensions don't ring. ******************************************************************************************************* LOG: Can't find user [some_user(a)location.main-domain.com] LOG: Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] ******************************************************************************************************* Our bridge isn't anything special, but I put it out here so that there's no question about what we're doing: bridge( [execute_on_originate=set_user::user1@location.main-domain.com]user/user1@location.main-domain.com, [execute_on_originate=set_user::user2@location.main-domain.com]user/user2@location.main-domain.com, [execute_on_originate=set_user::user3@location.main-domain.com]user/user3@location.main-domain.com, ) Anyway, some locations report that not all extensions are ringing. There doesn't seem to be a pattern about when it occurs or what extension it happens to. While it seems to be only a hand full of them, I suspect that it's under reported. I know the extensions are all registered and I've tried calling each directly and they work. Anyone have any ideas? Thanks, Gabe

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recording's channels shifted
by Tóth Róbert 20 Sep '12

20 Sep '12

Hi Why left and right channels' voice shifted from each other in the recorded mp3? I record the sessions' voice thios way: <action application="set" data="execute_on_answer=record_session $${base_dir}/Record/${strftime(%Y%m%d-%H%M%S)}.mp3"/> When i listen back the mp3, at begining the two chennels are synchonized, but after some minutes there will be a shifting between the channels. Why? thanks, Robert

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Demo IVR
by Brian West 20 Sep '12

20 Sep '12

This does actually ring a real person... aka ME. Please don't tell me 'fuck you' when you call. While I appreciate that... Its not fulfilling for me. This is setup as a test for people to call, I do like talking to people from around the world, its fun. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

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How to force FS to execute?
by Emrah 20 Sep '12

20 Sep '12

Hi all, My example below cannot work because FS does not store the value of ${pin} until the call goes into execution state. While keeping it simple, what would be your take on this? Is there a way to instruct FS to start executing at some point and discovering the dialplan along the way? <extension name="pin test"> <condition field="destination_number" expression="^111$" break="never"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="read" data="4 4 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav pin 10000 #,*" inline="true" /> </condition> <condition field="${pin}" expression="^1234$"> <action application="transfer" data ="3000 XML local" /> </condition> <condition field="${pin}" expression="^\d{4}$"> <action application="hangup" /> </condition> </extension> Thanks and all the best, as always. Emrah

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FreeSWITCH Conference Call - What Would You Like To Talk About?
by Michael Collins 19 Sep '12

19 Sep '12

Hey all, While I'm thinking about it I would like to ask you to give me some ideas on what we can discuss in our upcoming weekly conference calls. We have openings from early October through the end of the year. I have a few more lined up but I definitely would like to see some input from the community on what else we can do on Wednesdays. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Conference Call Today: Repro SIP proxy
by Michael Collins 19 Sep '12

19 Sep '12

Hello all, Please join us for the conference call at 1PM EDT, 10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19 Scott and Daniel will be discussing the Repro SIP proxy and the ReSIProcate SIP stack. Talk to you soon. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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ERROR: sofia.c:2491 Error Creating SIP UA for profile: internal
by Tiago N. Sampaio 19 Sep '12

19 Sep '12

Hi All! Im writing an application to manage FS via mod_xml_curl. Im generating sofia.conf just generating default configuration joining ../sip_profiles/* on my xml. Its an example: <?xml version="1.0" encoding="UTF8"?> <document type="freeswitch/xml"> <section name="configuration"> <configuration name="sofia.conf" description="Sessao de configuracao sofia.conf"> <global_settings> <log-level value="0"/> <debug-presence value="0"/> </global_settings> <profiles> <profile name="external"> <gateways> </gateways> <aliases/> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param debug="0"/> <param sip-trace="no"/> <param sip-capture="no"/> <param rfc2833-pt="101"/> <param sip-port="5080"/> <param dialplan="XML"/> <param context="public"/> <param dtmf-duration="2000"/> <param inbound-codec-prefs="$${global_codec_prefs}"/> <param outbound-codec-prefs="$${outbound_codec_prefs}"/> <param hold-music="$${hold_music}"/> <param rtp-timer-name="soft"/> <param local-network-acl="localnet.auto"/> <param manage-presence="auto"/> <param inbound-codec-negotiation="generous"/> <param nonce-ttl="60"/> <param auth-calls="false"/> <param rtp-ip="$${local_ip_v4}"/> <param sip-ip="$${local_ip_v4}"/> <param ext-rtp-ip="auto-nat"/> <param ext-sip-ip="auto-nat"/> <param rtp-timeout-sec="300"/> <param rtp-hold-timeout-sec="1800"/> <param tls="$${external_ssl_enable}"/> <param tls-bind-params="transport=tls"/> <param tls-sip-port="$${external_tls_port}"/> <param tls-cert-dir="$${external_ssl_dir}"/> <param tls-version="$${sip_tls_version}"/> </settings> </profile> <profile name="internal"> <aliases/> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param debug="0"/> <param sip-trace="no"/> <param sip-capture="no"/> <param watchdog-enabled="no"/> <param watchdog-step-timeout="30000"/> <param watchdog-event-timeout="30000"/> <param log-auth-failures="true"/> <param forward-unsolicited-mwi-notify="true"/> <param context="padrao"/> <param sip-port="$${internal_sip_port}"/> <param dialplan="XML"/> <param dtmf-duration="2000"/> <param inbound-codec-prefs="$${global_codec_prefs}"/> <param outbound-codec-prefs="$${outbound_codec_prefs}"/> <param hold-music="$${hold_music}"/> <param rtp-timer-name="soft"/> <param rtp-ip="$${local_ip_v4}"/> <param sip-ip="$${local_ip_v4}"/> <param apply-nat-acl="nat.auto"/> <param apply-inbound-acl="domain"/> <param local-network-acl="localnet.auto"/> <param presence-hosts="$${domain},$${local_ip_v4}"/> <param presence-privacy="$${presence_privacy}"/> <param inbound-codec-negotiation="generous"/> <param tls="$${internal_ssl_enable}"/> <param tls-bind-params="transport=tls"/> <param tls-sip-port="$${internal_tls_port}"/> <param tls-cert-dir="$${external_ssl_dir}"/> <param tls-version="$${sip_tls_version}"/> <param tls-verify-date="true"/> <param tls-verify-policy="none"/> <param tls-verify-depth="2"/> <param tls-verify-in-subjects=""/> <param inbound-late-negotiation="true"/> <param inbound-zrtp-passthru="true"/> <param nonce-ttl="60"/> <param auth-calls="$${internal_auth_calls}"/> <param inbound-reg-force-matching-username="true"/> <param auth-all-packets="false"/> <param ext-rtp-ip="auto-nat"/> <param ext-sip-ip="auto-nat"/> <param rtp-timeout-sec="300"/> <param rtp-hold-timeout-sec="1800"/> <param force-register-domain="$${domain}"/> <param force-subscription-domain="$${domain}"/> <param force-register-db-domain="$${domain}"/> <param challenge-realm="auto_from"/> </settings> </profile> </profiles> </configuration> </section> </document> but when i do reload mod_sofia i got this: 2012-09-19 11:42:52.548225 [CONSOLE] sofia.c:1620 MSG Thread Ended 2012-09-19 11:42:52.548225 [CONSOLE] switch_loadable_module.c:1814 mod_sofia unloaded. 2012-09-19 11:42:53.268217 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/41a9f34a-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.348238 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/c8a3004b-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.348238 [INFO] mod_enum.c:871 ENUM Reloaded 2012-09-19 11:42:53.428240 [INFO] mod_sofia.c:5652 Starting initial message thread. 2012-09-19 11:42:53.428240 [CONSOLE] sofia.c:1602 MSG Thread 0 Started 2012-09-19 11:42:53.488212 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/7fbd0d4b-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.488212 [INFO] switch_time.c:1163 Timezone reloaded 530 definitions 2012-09-19 11:42:53.568343 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/b333194b-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.588215 [INFO] sofia.c:5281 Setting nonce TTL to 60 seconds 2012-09-19 11:42:53.588215 [NOTICE] sofia.c:5358 Started Profile external [sofia_reg_external] 2012-09-19 11:42:53.668232 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/42f5314b-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.668232 [NOTICE] sofia_reg.c:2954 Added gateway 'OTIMA' to profile 'external' 2012-09-19 11:42:53.768237 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/8d68404b-6802-e211-9d5d-ced1711242b1.tmp.xml 2012-09-19 11:42:53.768237 [INFO] sofia.c:5281 Setting nonce TTL to 60 seconds 2012-09-19 11:42:53.768237 [NOTICE] sofia.c:5358 Started Profile internal [sofia_reg_internal] 2012-09-19 11:42:53.768237 [ERR] sofia.c:2491 Error Creating SIP UA for profile: internal 2012-09-19 11:42:54.768518 [NOTICE] sofia_reg.c:415 Registering OTIMA 2012-09-19 11:42:55.288223 [CONSOLE] switch_loadable_module.c:1328 Successfully Loaded [mod_sofia] 2012-09-19 11:42:55.288223 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'sofia' 2012-09-19 11:42:55.288223 [NOTICE] switch_loadable_module.c:254 Adding Application 'sofia_sla' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_gateway_data' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_username_of' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_contact' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_count_reg' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_dig' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sip' 2012-09-19 11:42:55.308234 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001] I searched a lot to get help, but cant find...

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ESL API command and ringback
by Peter van Raamsdonk 19 Sep '12

19 Sep '12

Hi again [?], Simple question, In the dialplan there is a Local_Extension to handle calls. In this example, you have an action; *<action application="set" data="ringback=${us-ring}"/>* As api command (from ESL java outbound example) I put this like; String result = doAction("global_getvar", "us-ring");// using CLI (inbound), result *%(2000,4000,440,480)* as defined in vars.xml SendMsg sendMsg = *new* SendMsg(); sendMsg.addCallCommand("execute"); sendMsg.addExecuteAppName("set"); sendMsg.addExecuteAppArg("ringback=" + result); but I don't hear sound. I don't know if the string argument "${us-string}", can be parsed directly. I tried everything but no sound. Only thing that I got working is by using "execute" and "ring_ready" What I'm doing wrong with ringback argument? Kind regards, Peter

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How to disable adding of rows with LOSE_RACE case ?
by Yuriy Nasida 19 Sep '12

19 Sep '12

Hello guys! I use many endpoints in my bridge appl. So in cdr I see one good row (for ext which picks up call) and many rows with "LOSE_RACE" for others ext from bridge string. How can I disable adding of rows with LOSE_RACE ? Please advice.Thanks.

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