December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Speed Dial Emulation
by jflowers 15 Dec '08

15 Dec '08

How do I emulate a speed dial setup. That is, from extension 1003 I dial just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a PSTN number. Is there software to do this? -- View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Multiple mod_portaudio channels
by Clifford 15 Dec '08

15 Dec '08

I need to set up a system that would allow multiple mod_portaudio channels to be configured. The calls would come in over SIP and any calls would be assigned a specific port on the multi-port sound card. I would probably set it up so specific extensions go to specific ports, for example 1001 would be port 1 and 1008 would be port 8 on the sound card. The documentation on the mod_portaudio module is very slim. In fact I would say it is nearly undocumented. Is it capable of handling multiple audio device ports (such as an 8-port LYNX sound card)? If so how would it be interfaced/configured? Thanks, Clifford

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Openzap ERROR can't dial
by Franziska Röhler 15 Dec '08

15 Dec '08

Hello, Im a newbie in FS and my English isnt very good but I try to explain my problem. Hopefully you can understand me. :-) I have a Linux-Server with a Digium Wildcard TE110P. I install and configure openzap (PRI/ISDN). But when I make an outbound call, I have an error. FS cant dial. [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] I have this warnings too, when no call is done: 2008-12-15 10:11:14 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 2008-12-15 10:11:15 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 7 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:1) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:2 (1:2) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:3 (1:3) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:4 (1:4) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:5 (1:5) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:6 (1:6) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:7 (1:7) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:8 (1:8) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:9 (1:9) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:10 (1:10) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:11 (1:11) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:12 (1:12) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:13 (1:13) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:14 (1:14) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:15 (1:15) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:16 (1:17) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:17 (1:18) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:18 (1:19) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:19 (1:20) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:20 (1:21) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:21 (1:22) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:22 (1:23) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:23 (1:24) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:24 (1:25) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:25 (1:26) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:26 (1:27) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:27 (1:28) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:28 (1:29) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:29 (1:30) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:30 (1:31) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:31 (1:16) has alarms [YELLOW] 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:1 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:2 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:3 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:4 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:5 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:6 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:7 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:8 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:9 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:10 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:11 What do I wrong? I hope you can help me! Here my configuration: OPENZAP.CONF [span zt] name => OpenZap number => 1 trunk_type => e1 b-channel => 1-15, 17-31 d-channel => 16 OPENZAP.CONF.XML <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> ZT.CONF [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 DEFAULT.XML <extension name="outgoing_pri"> <condition field="destination_number" expression="^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="openzap/1/A/${dialed_ext}"/> </condition> </extension> When I have forget to display some Configuration to , you can tell me!

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Bug in Q921.c? AVAYA-PBX issue
by Helmut Kuper 15 Dec '08

15 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04 server in a non-root environment. We experienced a timer problem which led to this FS console error message: [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries During anylizing this we found that q921 T203 is never reset when link is in state "Multiple Frame Mode Established" and SABME frames are received by FS. So it must timeout regardless if SABME frames are received or not. Additionally we found that the default T203 value (10 sec) was too short for AVAYA (it has to be >=19 sec) To fix the problem we changed two things in q921.c: Change T203 default value from 10 sec to 20000 sec Line 406: trunk->T203Timeout = 20000; Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each received SABME frame Line 1996: Q921T203TimerReset(trunk, tei); After recompiling FS the Error disapeared. Next week we will do some calls over the link to make sure there are no other side effects. Is it planned to make the q921 timeouts configurable in openzap.conf or in openzap.conf.xml? best regards Helmut PS: My openzap configs: openzap.conf [span wanpipe PRI_1] trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="7"/>    </settings> <pri_spans> <span name="PRI_1">  <param name="q921loglevel" value="debug"/> <param name="q931loglevel" value="debug"/> <param name="mode" value="user"/> <param name="dialect" value="Oh this is not my dialect"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> Very interesting here is, that the "dialect" parameter doesn't seem to have an effect on FS. I use that one above without any errors or warning and I guess that was not intended. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB =geXp -----END PGP SIGNATURE-----

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Maintaining call detail record
by Faisal Maqsoodi 15 Dec '08

15 Dec '08

if (typeof YAHOO == "undefined") { var YAHOO = {}; } YAHOO.Shortcuts = YAHOO.Shortcuts || {}; YAHOO.Shortcuts.hasSensitiveText = false; YAHOO.Shortcuts.sensitivityType = []; YAHOO.Shortcuts.doUlt = false; YAHOO.Shortcuts.location = "us"; YAHOO.Shortcuts.document_id = 0; YAHOO.Shortcuts.document_type = ""; YAHOO.Shortcuts.document_title = "[Freeswitch-users] Maintaining call detail record"; YAHOO.Shortcuts.document_publish_date = ""; YAHOO.Shortcuts.document_author = "faisalmaqsoodi(a)yahoo.com"; YAHOO.Shortcuts.document_url = ""; YAHOO.Shortcuts.document_tags = ""; YAHOO.Shortcuts.document_language = "english"; YAHOO.Shortcuts.annotationSet = { "lw_1229319648_0": { "text": "caller id", "extended": 0, "startchar": 188, "endchar": 196, "start": 188, "end": 196, "extendedFrom": "", "predictedCategory": "", "predictionProbability": "0", "weight": 0.2054, "relScore": 5.33095, "type": ["shortcuts:/concept"], "category": ["CONCEPT"], "wikiId": "Caller_ID", "relatedWikiIds": [], "relatedEntities": [], "showOnClick": [], "context": "fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked", "metaData": { "visible": "true" } }, "lw_1229319648_1": { "text": "http://wiki.freeswitch.org/wiki/Mod_cdr_csv,", "extended": 0, "startchar": 426, "endchar": 469, "start": 426, "end": 469, "extendedFrom": "", "predictedCategory": "", "predictionProbability": "0", "weight": 1, "relScore": 0, "type": ["shortcuts:/us/instance/identifier/URL"], "category": ["IDENTIFIER"], "wikiId": "", "relatedWikiIds": [], "relatedEntities": [], "showOnClick": [], "context": "", "metaData": { "visible": "true" } } }; YAHOO.Shortcuts.headerID = "284c8f98b4fb0aebc968053934caa66b"; How can i interface fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked on xml cdr but it contains too much info, more than i need and in a format which is not easily understandable. I also tried perl coding mentioned on the link at the bottom of the page http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me. faisal

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references to source files in error messages
by Jason White 14 Dec '08

14 Dec '08

I would just like to thank the FreeSWITCH developers for including the source file names and line numbers in error messages. This is not only helpful to the authors of the software, but to anyone who can read C code.

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Sip profiles used in bridge application
by Jason White 14 Dec '08

14 Dec '08

The Wiki page at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall gives the following example: <action application="bridge" data="sofia/sip/9998881111(a)sip.yourprovider.com"/> which, for me at least, doesn't work due to an invalid profile error: [ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile However, if I replace "sip" in the above action with "external" to specify the external profile, it works. I suspect this is a documentation error rather than a code bug. However, I couldn't find an explanation of the syntax, in particular, whether the profile name has to match a profile defined in the SIP configuration or whether there are profile names defined in the code that have special meanings as well. I'm using the default configuration for now. I could go through the code to find this out, of course, but I thought it better to ask here instead.

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How to control to domain used in INVITE From header
by Jon Bruel 14 Dec '08

14 Dec '08

The situation is as follows: An incoming call is processed by the FS and sent out to a sip client. I want to control the From header in this outgoing INVITE. I have tried to set various channel variables, including sip_h_From, in order to control to domain used in the INVITE From header, which for instance looks like this: From: "JBS (Soft)" <sip:205@X.X.X.X> Instead of the server IP address, X.X.X.X, I want to set the SIP-domain used for the specific customer in a multi tenant setup. This from header is used by the phone telephone number list register (at least for the Snom phones), so controlling it is important. How is it done? /Jon

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libtool version.
by henkoegema 14 Dec '08

14 Dec '08

root@MSI:/home/henkoegema/freeswitch# ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.61 (ok) bootstrap: automake version 1.10.1 (ok) bootstrap: libtool version 2.2.4 found. You need libtool version 1.5.14 or newer installed <------Isn't that what I have :confused: to build FreeSWITCH from SVN. root@MSI:/home/henkoegema/freeswitch# -- View this message in context: http://www.nabble.com/libtool-version.-tp21000027p21000027.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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schedule a DTMF tone into bridge
by Frank ＠ Impact 14 Dec '08

14 Dec '08

Is there a way to schedule a certain DTMF tone to be played into a bridge (both a and b legs) after a scheduled number of seconds into the call?

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