October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

german prompts - first release
by Christian Löschenkohl 08 Oct '10

08 Oct '10

hello freeswitch community i can now announce the first release of the german prompt set. you can download it under http://freeswitch.xpirio.com/. this package is not useable at this time, because there are some xml files that need to be created or changed first. also the prompts are not 100% perfect and complete. i'll work on these topics and keep you updated. if somebody wants to help, please take a look at the prompts an send me your corrections (directly to my e-mail address, not to the list please). also look at the file de2.xls where you can find the prompts the need to be recorded or corrected. please take a look at the xml files under conf/lang, they also need to be created and/or modified too. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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FreeTDM Partial Spans?
by Jeroen C. van Gelderen 08 Oct '10

08 Oct '10

Hi Guys, This might be a dumb question but I can't seem to find an answer by Googling. I have oodles of FXO ports configured on Xorcom Astribanks. On the Astribank each set of 8 FXO ports forms a zaptel/openzap/freetdm span. Now if a trunk consists of an integral multiple of 8 FXO ports it is easy to just dial out on the first available channel: <action application="bridge" data="freetdm/1/a/$1|freetdm/2/a/$1"/> This grabs the first available channel out of 16 FXO ports. But what is the "right" way to do it when I need to use 1.5 spans (i.e. 12 FXO ports)? Is there such a thing as a virtual span that can be built out of individual FXO ports? I have a feeling I'm missing something rather obvious. Thanks, -Slim -- Jeroen C. "Slim" van Gelderen Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: slim(a)thegreek.com Phone: +1 876 953 6182 x128

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Loss of first second of media from termination provider
by Fraser Redmond 08 Oct '10

08 Oct '10

I've just set up with a new termination provider, and am finding that the first second of media is being lost. I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine. Their engineers have run a sip trace and a full packet trace, and it looks like everything is being delivered from the pstn to my server. They said that none of their other customers have reported this problem before. I've tested connecting to pstn's in several different countries, with the same results. I've used 2-3 other termination providers (1 still in current use) which have all been fine. So it seems like it's a combination of Freeswitch with this provider. This is running on Ubuntu on a Amazon AWS server, the version of freeswitch is about 2 weeks old, I'll be trying on a new server with the latest version tomorrow, in case that helps. Any ideas? Cheers, Fraser

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mod_dingaling segfault with libgcrypt and gnutls on Fedora 13
by Jeffrey Leung 08 Oct '10

08 Oct '10

Hey all, I've read and understand somewhere that with a bad copy of gnutls and libgcrypt can cause mod_dingaling to segfault when loading or running. However, I'd like to ask if there's anyone out there trying to use mod_dingaling with Fedora 13's included libgcrypt and gnutls-devel packages as I am experiencing segfaults when I try to load mod_dingaling.

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FreeSWITCH parameters for voice quality
by rex.alex 08 Oct '10

08 Oct '10

Hello, I have FreeSWITCH on Cent OS 5.4 physical server and I am using X-Lite soft phone on Windows client to make external calling through a registered gateway. But, I am facing issues like voice breakage and static noise on calls. I am monitoring the network connectivity but before that I want to make sure that all FreeSWITCH parameters have been configured correctly. Please let me know as far as FreeSWITCH is concern what are the parameters that I have to check for these kind of issues? Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-parameters-for-voi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch - mod_spidermonkey_curl bug?
by Shamun toha md 08 Oct '10

08 Oct '10

Hello, I have a very strange problem using mod_spidermonkey_curl in CentOS 5.5 x86_64 bit. Same setup i am using in other machine in Fedora 12, it works, is there any bugs????????? I receive OK, but when i use to compare it fails in logical statement. FreeSWITCH Version 1.0.head (git-25ddef5 2010-09-24 01-55-54 +0200) Started. Installation curl as: ============== [root@a autoload_configs]# cat spidermonkey.conf.xml <configuration name="spidermonkey.conf" description="Spider Monkey JavaScript Plug-Ins"> <modules> <load module="mod_spidermonkey_teletone"/> <load module="mod_spidermonkey_core_db"/> <load module="mod_spidermonkey_socket"/>  * <load module="mod_spidermonkey_curl" />* </modules> </configuration> .../scripts/helloworld.js: ============ use("CURL"); var menuselection = ""; var promptfordigits_dtmf_digits = ""; var invalidRetry = 0; /* This is getting failed, in Centos, but in other machine it works */ function my_callback(string, arg) { console_log("info", string); //string = 'OK'; // then it works!! //2010-10-08 18:40:53.585676 [INFO] mod_dialplan_xml.c:331 Processing 1002 <1002>->1200 in context default * //2010-10-08 18:41:23.842965 [INFO] helloworld.js:42 OK* //2010-10-08 18:41:35.962099 [NOTICE] switch_ivr.c:1480 Transfer sofia/internal/1002(a)z.com to XML[3000@default] //2010-10-08 18:41:35.962099 [INFO] mod_dialplan_xml.c:331 Processing 1002 <1002>->3000 in context default //2010-10-08 18:41:43.134207 [NOTICE] sofia.c:528 Hangup sofia/internal/ z.com [CS_EXECUTE] [NORMAL_CLEARING] if (string=='OK') { // // // <<<<<<<<<<<<<<<<<<------------------ It receive OK but it never execute this two, and but it execute invalid macro // // // session.sayPhrase("valid", menuselection, "en"); session.execute("transfer", '3000'); } else { session.sayPhrase("invalid", menuselection, "en"); return true; } /* FS wiki - example */ function promptfordigits_dtmf_callback(session, type, digits, arg) { console_log("digit: " + digits.digit + "\n"); promptfordigits_dtmf_digits += digits.digit; /* returning true does not interrupt the digit collection*/ return(true); } /* FS wiki - example */ function promptfordigits(ivrsession, promptname, numdigits, timeout) { var repeat = 0; console_log("saymenu: menu=[" + promptname + "] numdigits=[" + numdigits + "]\n"); session.flushDigits(); promptfordigits_dtmf_digits = ""; while (ivrsession.ready() && promptfordigits_dtmf_digits.length < numdigits && repeat < 3) { /* play phrase - if digit keyed while playing callback will catch them. If less than numdigits collected we get the rest after the prompt.*/ ivrsession.sayPhrase(promptname, numdigits, "en", promptfordigits_dtmf_callback, ""); console_log("Prompt done=[" + promptname + "] Collected " + promptfordigits_dtmf_digits.length + " digits [" + promptfordigits_dtmf_digits + "]\n"); /* if caller still here and has not entered any selection yet (or less than numdigits entered) - wait for the rest of the digits*/ if (ivrsession.ready() && promptfordigits_dtmf_digits.length < numdigits ) { promptfordigits_dtmf_digits += ivrsession.getDigits(numdigits - promptfordigits_dtmf_digits.length, "", timeout); /* if still no selection or insufficient digits repeat menu */ if (promptfordigits_dtmf_digits.length < numdigits) { promptfordigits_dtmf_digits = ""; repeat++; } } } return(promptfordigits_dtmf_digits); } function retry() { /* FS wiki - example */ menuselection = promptfordigits(session, "DialByNumberMenu", 8, 6000); if ( session.ready() ) { var curl = new CURL(); curl.run( "POST", "http://portal/conference/query", "om=" + menuselection + "&ty=__", my_callback, "my arg\n", "p:ier"); } } /** * Let's answer our call */ session.answer(); retry(); Thanks & Regards

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FS starts re-registering every minute after a few hours
by Mario 08 Oct '10

08 Oct '10

I really could use some help on this one, this is one of the last major issues before I switch to FS. I have expire-seconds set at 600, this works for several hours. Then, for no apparent reason it changes to 60 (see trace). Most calls are missed due to this issue. Why would FS change it and how can I keep it at 600? BTW retry is at 30. I spent lot's of time with traces/investigating and the ITSP watched it and says they are not changing anything. Thanks! Mario PS If it's them how do I prove it? send 865 bytes to udp/[111.222.1.36]:5060 at 15:07:22.220308: ------------------------------------------------------------------------ REGISTER sip:sip.itsp1.com SIP/2.0 Via: SIP/2.0/UDP 100.101.102.103:5080;rport;branch=hideBcm183mmt2UD Max-Forwards: 70 From: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=hide37BeUD To: <sip:uuid1@sip.itsp1.com;transport=udp> Call-ID: hide-a069-49d0-b342-hide34d28 CSeq: 2914438 REGISTER Contact: <sip:gw+uuid1@100.101.102.103:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f499c0c 2010-10-07 10-27-31 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="xxxtel.com", nonce="hide0000123270bed298a5fee464c4bb367750hide", algorithm=MD5, uri="sip:sip.itsp1.com", response="70000a710c1b7a363899922e6hide" Content-Length: 0 ------------------------------------------------------------------------ recv 567 bytes from udp/[111.222.1.36]:5060 at 15:07:22.321019: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 100.101.102.103:5080;rport=5080;branch=hideBcm183mmt2UD;received=111.222.333.127 From: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=hide37BeUD To: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=9214944d68c027e4c99a14a397962871.4d99 Call-ID: hide-a069-49d0-b342-hide34d28 CSeq: 2914438 REGISTER Proxy-Authenticate: Digest realm="xxxtel.com", nonce="4caf33ab99012dadc6e7c84040e1b4ff8841a343edc5b0c3", stale=true Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ send 865 bytes to udp/[111.222.1.36]:5060 at 15:07:22.321391: ------------------------------------------------------------------------ REGISTER sip:sip.itsp1.com SIP/2.0 Via: SIP/2.0/UDP 100.101.102.103:5080;rport;branch=z9hG4b99m5c33mrH3reS Max-Forwards: 70 From: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=hide37BeUD To: <sip:uuid1@sip.itsp1.com;transport=udp> Call-ID: hide-a069-49d0-b342-hide34d28 CSeq: 2914439 REGISTER Contact: <sip:gw+uuid1@100.101.102.103:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f499c0c 2010-10-07 10-27-31 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="xxxtel.com", nonce="4caf33ab00012dadc6e7c89940e1b4ff9941a343edc5b0c3", algorithm=MD5, uri="sip:sip.itsp1.com", response="b8597203297d6db518b9996f55d3dc36" Content-Length: 0 ------------------------------------------------------------------------ recv 503 bytes from udp/[111.222.1.36]:5060 at 15:07:22.421206: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 100.101.102.103:5080;rport=5080;branch=z9hG4bKKm5c999rH3reS;received=111.222.333.127 From: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=hide37BeUD To: <sip:uuid1@sip.itsp1.com;transport=udp>;tag=921495599c027e4c99a14a397962871.9262 Call-ID: hide-a069-49d0-b342-hide34d28 CSeq: 2914439 REGISTER Contact: <sip:gw+uuid1@111.222.333.127:5080;transport=udp;gw=uuid1>;expires=60 Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0

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bypass_media only if both peers have public IP: how to do it?
by Fabio Pietrosanti (naif) 08 Oct '10

08 Oct '10

Hi, i would like to configure a FS system so that only if both peers have a public IP address they can speak each other. The condition is that the client are mobile and can connect from whatever network, so apply-nat-acl does not work because some mobile carrier use fancy internal IP address such as 1.1.1.1 . All clients connect with SIP/TLS and there is no SIP ALG that mangle the SDP, so we can be sure that what FS see is exactly what the SIP client has sent. Given that, i would like FS to always do proxy_media unless BOTH two leg of the call have a public IP address. I know that's not extremely efficient and that in theory two SIP clients with one behind NAT and one with public IP can speak each other, but even in such conditions there may be unpredictable issues. How the NAT handling of FS could be modified to do that and let bypass_media only when both SIP client have a public IP address? Fabio

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segfault in swift library
by covici＠ccs.covici.com 08 Oct '10

08 Oct '10

I have been trying to get mod_cepstral to work and so far, no joy at all. At first I was using the version 5 voice, but was told on the IRC or maybe in the onference that I should go back to version 4. So I restore the directory, recompiled FSW and now I see in the logs the following message: Oct 7 06:16:15 ccs kernel: freeswitch[10398]: segfault at 627769 ip b5ffda43 sp bfd48ce4 error 4 in libswift.so.4.2[b5fcf000+7e000] No core was dumped -- can someone tell me what is going on, or how to troubleshoot this? Thanks in advance for any suggestions. -- 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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Possible bug with bypass_media=true functionality
by Pete Kelly 08 Oct '10

08 Oct '10

Hi, I posted a question to this list yesterday because I was having problems with my calls ending as soon as they were bridged. I was originating 2 outbound calls from a lua script, then bridging them. I managed to get round the problem of the calls shutting down immediately by initiating calls using the originate command, thusly: originate sofia/lpmedia/2000(a)10.15.20.122 '&lua(click2call_call.lua)' This causes the lua script to initiate with a session already present and then a bridge to a new call (legB) completes successfully. Within the lua script I am bridging to leg B like this: session:execute("bridge", "[bypass_media_after_bridge=true]sofia/lpmedia/ 91979197(a)10.15.20.122") This works, but the reINVITEs which get sent out after legB is established contain the wrong sip request URI. - The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the SIP packet is sent to the correct IP, just the request URI contains the media IP) - The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A... again the SIP packet is sent to the correct IP, it's just the request URI which contains the media IP. In some instances the request URI doesn't contain the username either, it is just INVITE sip:<MEDIA_IP> I am using the latest sources of freeswitch from git Is this a bug I need to report? It is causing me to receive 404 and 402 errors on the reinvite as the request URI is invalid. Pete

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