July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

No audio when calling in via SIP phone
by Iqbal Abdullah 30 Jul '14

30 Jul '14

Hello everyone, I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system. My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/ via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead. One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu. This is the log on freeswitch: 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [CS_DESTROY] I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server: [29/Jul/2014 01:59:13] "GET /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93 This is the XML I have prepared to test plivo: <?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response> And this is default.xml on freeswitch which forward any calls to 1005 to plivo: <include> <context name="default">  <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include> I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying: 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. but what does that mean?

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Avoiding media timeouts in remote systems
by Richard Brady 30 Jul '14

30 Jul '14

I have FS talking to a carrier, and got on-hold the carrier sends a reINVITE with a=sendonly, so FS stops sending media, but the carrier drops the call due to a media timeout. Appears their SIP and RTP stacks aren't talking to each other very well. Can I compensate for this by continuing to send RTP to keep the their MGW session alive? Regards, Richard

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fax receiving issue (two page compressed into one)
by Marlok Tamas 30 Jul '14

30 Jul '14

Hi, I have the following problem: There are a few fax number from where I all two page faxes are compressed into one page (see the attached pdf). I could not find anything about this issue so far... The log, and the received pdf (generated from the tiff - the problem is not in the tiff-pdf conversion): https://www.dropbox.com/s/vxkjygxyy5zi9d6/log.txt https://www.dropbox.com/s/49udrprgmascc0x/test1.pdf The relevant part of the receiver python script is: the_uuid = session.getVariable("uuid") the_recipient = session.getVariable("recipient") the_caller = session.getVariable("caller_id_number") the_dest = re.sub(r'^(\+?3|0)(6[0-9]*)$', r'3\2',session.getVariable("destination_number")) send_from = the_dest+'@fax.voip123.hu' consoleLog("info", " rxfax receiving from " + the_caller + " for destination " + the_dest + ", uuid is " + the_uuid + "\n") fname = incomingfaxes + "rxfax-" + the_uuid #answer the phone, save the fax file in .tiff format. session.answer() session.execute ("playback", "silence_stream://2000") # session.execute ("set","fax_enable_t38=true") session.execute ("rxfax", fname + ".tiff") Thanks in advance! Cheers, tmarlok

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Provisioning through extension
by Fernando Flórez 30 Jul '14

30 Jul '14

Hello, I would love to have some guidance on how to achieve user registration through an extension. Phones will be placed on a restricted context by default, if a user calls an specific extension they will be prompted to enter their extension and password and if successful, the phone will start working under that extension on a different (not restricted) context. Multiple phone registration should also be possible. Thank you!

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Freeswitch Week in Review(Master Branch) July 20th-26th
by Kathleen King 30 Jul '14

30 Jul '14

Hello, again. This week in the FreeSWITCH master branch we had 58 commits. Some of the new features that were added include more tweaks and improvements to mod_verto, the addition of a final_delivery variable to the chatplan to prevent delivery of pre-empted messages, and the new module mod_prefix an in-memory data store optimized for fast lookups. The following bugs were squashed: ac265ce FS-6682: fix arg order on switch_channel_export_variable called from switch_channel_export_variable_printf #resolve Jira: http://jira.freeswitch.org/browse/FS-6682 d9eafdf FS-6681: fix dup of audio/wav and add audio/x-wav Jira: http://jira.freeswitch.org/browse/FS-6681 025e65a Fix resampler issue caused by refactoring impacting audio quality of resampled calls in mod_conference. 5732571 FS-6689 Fix for package building issue in util.sh. Jira: http://jira.freeswitch.org/browse/FS-6689 3d5d816 Changes by Ryan Lantzer <lantzer(a)mst.edu> to correct the generation of line_instance value in outbound ringer requests in mod_skinny. 940171c FS-6670 Fixed a global function pointer problem in mod_sofia. Jira: http://jira.freeswitch.org/browse/FS-6670 cf786b1 FS-6685 Fix behavior of ring if other SLA extension is active in mod_skinny. Jira: http://jira.freeswitch.org/browse/FS-6685 c15663c FS-6670 Fixed global function pointer problem in libsofia. Jira: http://jira.freeswitch.org/browse/FS-6670 6eda1c0 Tweak timeout a little as it was changing to 1000000 on the 5th try on return from switch_interval_time_from_timeval on OpenBSD. New features that were added: 7596689 Added chat to mod_verto demo. acffa79 Send DTMF from keyboard strokes in mod_verto. cb7a338 Added back the jquery CDN into tree. fbf4ccd FS-6686 #resolve #comment increased the timeout to 30 to accommodate situations where the engine takes longer to pick an IP in mod_verto. Jira: http://jira.freeswitch.org/browse/FS-6686 c735f5c Add enter press detection to the extension field to auto-dial in mod_verto bbcbb24 Additional changes to mod_verto. e09ea08 Hack for dropbox in mod_verto. a47dc4b Render image URLs in mod_verto. c411f8c Queue more messages in mod_verto. 4f1b09a Add switch_socket_waitlist for waiting on multiple sockets and abstract it to poll and select so windows can use it too. bd804df Only do kepress on outside of page for dtmf in mod_verto. a537d80 Add new module: mod_prefix. a4ff7ea Render image URLs in mod_verto. 6f800ef Add resize in mod_verto. 7c01f06 Use final_delivery variable in chat plan to indicate that you do not want to deliver the message to its intended recipient because you have already pre-empted the message. Improvements in cross platform build supports: a1d73df Fix configure check for libjpeg to not break other checks. 407fb1a Enable libspandsp to build on openbsd. 31f98ef Misc work for the magic bsd makefile. 00573a9 Tweaks to build for openbsd. aaddc42 Extra measures to ensure use of local ssl library. 5ea1729 FS-6674 Fixed mod_verto failing to build in a non-source directory in mod_verto. Jira: http://jira.freeswitch.org/browse/FS-6674 ad8c7bc Added Curl build so it links against the correct ssl lib for openbsd. 78c07db Use LibreSSL on OpenBSD build. ff245ec Fix has-get for openbsd. 327146c Fix WS Compile on MSVC2012 35eec37 Added sounds to default make install target for openbsd, solaris11, and centos6. In terms of stability these were the use cases that were fixed: 1be1219 FS-6680 Fix for segfault upon startup on OpenBSD. Jira: http://jira.freeswitch.org/browse/FS-6680 7e08d21 FS-6663 Fixed a memory leak in mod_event_multicast. Jira: http://jira.freeswitch.org/browse/FS-6663 f11c40d FS-6680 Fix for segfault upon startup on OpenBSD. Jira: http://jira.freeswitch.org/browse/FS-6680 Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (253) 617-3210 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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ClueCon Weekly This Week The ClueCon Pregame Show!
by Ken Rice 29 Jul '14

29 Jul '14

Hey FreeSWITCHers and ClueCon¹ers! Join us for ClueCon Weekly Wed 1PM Eastern (1700 GMT) and get the latest updates on ClueCon and little sneak peak at what all is going on this year! We¹ll be at http://conference.freeswitch.org/ via WebRTC (use Chrome for the Stereo Goodness) or call us via the normal mechanism! If you havent registered for ClueCon there is still time! Visit ClueCon.com and register today! If you want to stay for the FreeSWITCH Training and have not yet registered get in there and register now! There is only 3 slots for training remaining! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] multiple languages
by Neo Haux 29 Jul '14

29 Jul '14

Hi Darcy, Merci beaucoup ! Before diving in this and try to implement these codes (for a single domain that I have) I want to be sure that I have the correct FS's setup. Do I need cepstral installed to be able to have voicemail in French? Because this config is for my home pbx/voicemail, so I am not sure if I'll pay the license for such a small setup. Thanks again. ForwardedMessage.eml Subject: Re: [Freeswitch-users] multiple languages From: "Darcy Primrose" <fs(a)voice2net.ca> Date: 07/28/2014 04:46 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> On more item on this that makes it simpler, in the directory you can set the variable <variable name="default_language" value="fr"/> Then just use one code {we use *98) to check voicemail. Justed tested this and it does work also. So I guess this makes it seamless. Darcy ----- Original Message ----- *From:* Darcy Primrose <mailto:fs@voice2net.ca> *To:* FreeSWITCH Users Help <mailto:freeswitch-users@lists.freeswitch.org> *Sent:* Monday, July 28, 2014 4:40 PM *Subject:* Re: [Freeswitch-users] multiple languages I inserted it in three places. First for the IVR, following is a portion of the dialplan that calls the ivr <condition field="${are_we_open}" expression="^Yes$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="default_language=fr"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="ivr" data="61_domain.mydomain.ca"/> <action application="transfer" data="820 XML domain.mydomain.ca"/> <anti-action application="transfer" data="20 XML domain.mydomain"/> </condition> So I use two IVRs, one for english, 1 for french. Then for inbound voice mail - this is for extension 28 in its dialplan <action application="bridge" data="{sip_invite_domain=domain.mydomain.ca,presence_id=28@domain.mydomain.ca}user/28@domain.mydomain.ca"/> <action application="set" data="default_language=fr"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default domain.mydomain.ca 28"/> Then for checking voicemail we use two codes currently, I haven't investigated this far enough to make it seamless, we had a customer to get going: <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4001$|^\*99$"> <action application="set" data="default_language=fr"/> <action application="answer"/> <action application="set" data="vm_announce_cid=ivr/ivr-this_is_a_call_from.wav"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default domain.mydomain.ca ${sip_from_user}"/> </condition> </extension> But the real key thing is to follow the instructions 100% on the following link, http://wiki.freeswitch.org/wiki/Mod_cepstral#libswift.so. And you have to purchase licenses to make it fully functional, but you get a courtesy make to let you know you have the correct language working. It is not 100% multilingual, but it is by all means acceptable. Hope this helps some. Darcy

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New FS crashed
by Oleg Stolyar 29 Jul '14

29 Jul '14

Hey guys, a shot in the dark here. I upgraded to the new master last week FreeSWITCH Version 1.5.14b+git~20140722T201952Z~35eec3704c~64bit (git 35eec37 2014-07-22 20:19:52Z 64bit) It crashed less than 24 hours after starting to take live calls. I don't have the core dump and I cannot reproduce it without putting the server back in production which of course would result in live calls being lost. So my questions are: 1. Is this a known issue by any chance? 2. If so, was it fixed in the latest master in the last week?

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P-Asserted-Identity issue
by Ričardas Stoma 29 Jul '14

29 Jul '14

Hello guys, i have a problem when i try to read P-Asserted-Identity header. Let's say i send to freeswitch this header: P-Asserted-Identity: <sip:123456789 at 192.168.0.73 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060;user=phone> But when i try to read this header in my dialplan, i get: 123456789 When i check channel variables i see: variable_sip_P-Asserted-Identity: 370370370370 maybe there is a way to read full P-Asserted-Identity header? I need to resend this header without modifying it. Is it possible?

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SSL cert required fields
by sangdrax8 29 Jul '14

29 Jul '14

I can get my latest freeswitch to run TLS profiles if I use the provided scripts to genenrate the CA and the server certs, but not with my own certs. The TLS profile does start with my cert, but when I connect, it does not provide a cert to the client. The only differences I can see in the server cert are the following fields: X509v3 Authority Key Identifier: DirName:/CN=FreeSWITCH CA/O=FreeSWITCH serial:91:F9:22:5D:22:38:6B:09 X509v3 Subject Alternative Name: DNS:test.freeswitch.org Netscape Cert Type: SSL Server X509v3 Extended Key Usage: TLS Web Server Authentication As I understand it, the Netscape and Usage designations are not used by freeswitch at this time. So I wouldn't expect them to cause an issue, unless this has changed since the documentation was written. I have the x509v3 Authority Key Identifier, with a keyid field, but I don't have these other two fields which the script puts. Are these required? I also do not have a Alt name defined in my cert, but I wouldn't have thought this would be a required field either. I can't find anything else that appears different to me. If someone can clarify what is required for Freeswitch to use a cert, it would be greatly appreciated!

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