July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Error in compilation after git pull
by aahndee 05 Jul '12

05 Jul '12

> Hi All, > > I did a git pull just now. After make clean, the compile failed. > > The latest git commit log that I can see is: > > commit 04bd463d12bf4c0116bec3cee537749b2040ed40 > Author: Michael S Collins <msc@...> > Date: Wed Jul 4 12:09:32 2012 -0700 > > The relevant error is: > > making all mod_spandsp > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, > from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. > h:33: > error: expected specifier-qualifier-list before > 'ademco_contactid_report_func_t' > cc1: warnings being treated as errors > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. > h:48: > error: struct has no members > make[4]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[3]: *** [mod_spandsp-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Am I missing something? I was able to compile a few weeks ago. > > Regards It seems like there are two header files for ademco_contactid.h: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h and /usr/local/src/freeswitch/libs/spandsp/src/spandsp/ademco_contactid.h 'ademco_contactid_report_func_t' is defined in the latter. I tried adding this second header file in /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h (before the private one) and managed to get a little further - now things fail with quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp_modem.lo -MD -MP -MF .deps/mod_spandsp_la -mod_spandsp_modem.Tpo -c mod_spandsp_modem.c -o mod_spandsp_la-mod_spandsp_modem.o >/dev/null 2>&1 make[5]: *** No rule to make target `config/ax_compiler_vendor.m4', needed by `Makefile.in'. Stop. make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2

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conference auto dial no video
by Seven Du 05 Jul '12

05 Jul '12

Hi, Is it possible to setup video when auto_call in conference? I see no video even I set absolute_codec_string <action application="set" data="conference_auto_outcall_prefix={absolute_codec_string=PCMA,PCMU,H264}"/> <action application="conference_set_auto_outcall" data="user/1000"/> <action application="conference" data="3000+flags{endconf}"/> INVITE sip:1000@10.64.14.206:42894;rinstance=c688826a567bedcb SIP/2.0 Via: SIP/2.0/UDP 10.64.14.206;rport;branch=z9hG4bKm16gttgFX1rrg Max-Forwards: 70 From: "FreeSWITCH" <sip:0000000000@10.64.14.206>;tag=Q49S334vvyBQS To: <sip:1000@10.64.14.206:42894;rinstance=c688826a567bedcb> Call-ID: 9ded3052-e523-49b5-949e-ec18ab13e0af CSeq: 30415873 INVITE Contact: <sip:mod_sofia@10.64.14.206:5060> User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120703T221514Z~b1ae97466d+unclean~20120704T022616Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 201 X-FS-Support: update_display,send_info Remote-Party-ID: "FreeSWITCH" <sip:0000000000@10.64.14.206>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1341480903 1341480904 IN IP4 10.64.14.206 s=FreeSWITCH c=IN IP4 10.64.14.206 t=0 0 m=audio 22972 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Thanks -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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Monitoring Success of SIP Registration
by Rob Moore 05 Jul '12

05 Jul '12

Hi Everyone, Recently we have had some problems with our Freeswitch boxes failing to respond to registration requests. A restart of FS always clears this up, and I'm looking else where for a long term solutions to this problem. However this does leave me with another situation while I look for the permanent solution. Its normally during out of hours that our freeswitch boxes decide they do not want to respond to registrations and I would like to setup some rolling registrations every 2 minutes that the result of can be reported back on to allow for some Alerting to take place. This could be from another Freeswitch server (maybe using a gateway, or some API commands run in a lua script) or a small application of some sort hosted on our monitoring platform. I even considered using SIP-P but everything I find seems to be rather heavy weight for what needs to be something a simple as a few lines of code in a lua script that returns a result. So I thought I 'd open it up to the users group to see if you had any suggestions on a light weight method I could use to generate these registration request. Thanks Rob

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Re: [Freeswitch-users] Wrong date/time in cli and events
by Peter Olsson 05 Jul '12

05 Jul '12

You just happened to be one of the few unlucky :) /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För David Notivol Skickat: den 5 juli 2012 13:39 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Wrong date/time in cli and events Thanks Peter, that solved the problem. I was running yesterday's git head, so I didn't think it was fixed since then. Regards, David Notivol. 2012/7/5 Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> Update to latest git, it was fixed yesterday. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För David Notivol Skickat: den 5 juli 2012 11:36 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: [Freeswitch-users] Wrong date/time in cli and events Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch@internal> strftime 1970-01-05 13:42:15 freeswitch@internal> strftime_tz 1970-01-05 freeswitch@internal> strepoch 412943 [root@hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff57bb032769674818857!

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Re: [Freeswitch-users] Wrong date/time in cli and events
by Peter Olsson 05 Jul '12

05 Jul '12

Update to latest git, it was fixed yesterday. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För David Notivol Skickat: den 5 juli 2012 11:36 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Wrong date/time in cli and events Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch@internal> strftime 1970-01-05 13:42:15 freeswitch@internal> strftime_tz 1970-01-05 freeswitch@internal> strepoch 412943 [root@hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol !DSPAM:4ff55f0732761568247525!

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Outbound ending after 30secs
by AP 05 Jul '12

05 Jul '12

Ok, so I have calls cutting off after 30 seconds on one type of handset but not another. Looking at ngrep, two things that jump out at me: 1. The working handset is sending ACK responses to FreeSWITCH's SDP messages, but the other one isn't. Presumably this is why FS is issuing BYE and dropping the call after 30 seconds, but there are no settings on the phone to this affect (Siemens Gigaset), and I'm not sure whether it's related to point 2 below: 2. The one which is cutting off is using UDP, and the working one TCP. TBH I can't see why it's using UDP when the SIP profile etc is set to use TCP!? Can anyone give me some pointers / see anything I'm missing? U 2012/07/03 15:40:18.370502 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. To: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" <sip:01234567890@(FS_PRIVATE_IP)>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # T 2012/07/03 15:40:19.841678 (ITSP_IP):5060 -> (FS_PRIVATE_IP):35664 [AP] OPTIONS sip:gw+MyGateway(FS_PUBLIC_IP):36304;tport=tcp;transport=tcp;gw=MyGateway SIP/2.0. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c. Max-Forwards: 70. From: <sip:mod_sofia@(ITSP_IP):5060>;tag=Fp3vKpcZBN4UF. To: <sip:istp_username@sip.my-itsp.com>. Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: <sip:mod_sofia@(ITSP_IP):5060>. User-Agent: my-itsp.com. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . # T 2012/07/03 15:40:19.842080 (FS_PRIVATE_IP):35664 -> (ITSP_IP):5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c;rport=5060. From: <sip:mod_sofia@(ITSP_IP):5060>;tag=Fp3vKpcZBN4UF. To: <sip:istp_username@sip.my-itsp.com>;tag=Q1y1N4BZU22pe. Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: <sip:gw+MyGateway(FS_PUBLIC_IP):5080;tport=tcp;transport=tcp;gw=MyGateway>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . ## U 2012/07/03 15:40:22.370503 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. To: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" <sip:01234567890@(FS_PRIVATE_IP)>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U 2012/07/03 15:40:22.863571 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 BYE sip:1002@(HANDSET_IP):5060 SIP/2.0. Via: SIP/2.0/UDP (FS_PUBLIC_IP);rport;branch=z9hG4bKp90ccpcU3jUaN. Max-Forwards: 70. From: <sip:01234567890@(FS_PRIVATE_IP);user=phone>;tag=graNreK1vcrXg. To: <sip:1002@(FS_PRIVATE_IP)>;tag=2284500536. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 30327147 BYE. Contact: <sip:01234567890@(FS_PUBLIC_IP):5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . (FS_PRIVATE_IP)

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Wrong date/time in cli and events
by David Notivol 05 Jul '12

05 Jul '12

Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch@internal> strftime 1970-01-05 13:42:15 freeswitch@internal> strftime_tz 1970-01-05 freeswitch@internal> strepoch 412943 [root@hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol

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Link to JIRA from the RSS feed in google reader? (greasemonkey help)
by Avi Marcus 05 Jul '12

05 Jul '12

Hi - I try to follow the commits to GIT - I use the RSS feed jira provides and follow in google reader. it's at http://fisheye.freeswitch.org/changelog/freeswitch.git?view=all&max=30&RSS=… Anyway - I try to see the actual issue to get context, but for example: FS-4381 isn't clickable... Can someone whip together a quick greasemonkey script (or patch jira!) to replace (FS-\d{4}) with a link to http://jira.freeswitch.org/browse/$1 I tried for about a half hour but couldn't even replace "FS" within a google reader window.... no clue what I'm doing. Thanks! -Avi

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Crash on incoming E1 call
by Robin Gilks 05 Jul '12

05 Jul '12

Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 This occurs on all incoming calls from a Digium Wildcard TE210P with a zaptel E1 interface with libpri. I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but failed miserably - couldn't even get the E1 link itself to come up :( I've captured a backtrace of the event on the zaptel drivers, hopefully useful to someone. gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands $ cat gdbcommands handle SIGPIPE nostop noprint handle SIG33 nostop noprint set logging on set pagination off set breakpoint pending on run thread apply all bt full set logging off Attached is the output of the backtrace. -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. It is subject to copyright, is confidential and may be the subject of legal or other privilege, none of which is waived or lost by reason of this transmission. If you are not an intended recipient, you may not use, disseminate, distribute or reproduce such email, any attachments, or any part thereof. If you have received a message in error, please notify the sender immediately and erase all copies of the message and any attachments. Unfortunately, we cannot warrant that the email has not been altered or corrupted during transmission nor can we guarantee that any email or any attachments are free from computer viruses or other conditions which may damage or interfere with recipient data, hardware or software. The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------

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how to capture packets of siptrace?
by Jack Nikolas 05 Jul '12

05 Jul '12

hi all, in conf.back/sip_profiles/internal.xml configuration i have enabled two options as below: <param name="sip-trace" value="yes"/> <param name="sip-capture" value="yes"/> but how can i capture packets when user is want to register sip client or making calls or logout or transfer, etc ? i want the following content ? recv 529 bytes from udp/[85.15.7.6]:23944 at 08:48:58.358101: SUBSCRIBE sip:XXXXXX Via: SIP/2.0/UDP XXXX:23944;branch=z9hG4bK-d8754z-7d25245fc51dd824-1---d8754z-;rport Max-Forwards: 70 Contact: XXXX To: XXXXX From: XXXXX Call-ID: ZjE3ZGMyNTBhODI5NTU0Y2RmZmMwYTY4ODQ2NDJjNzQ. CSeq: 4 SUBSCRIBE Expires: 0 User-Agent: eyeBeam release 1102q stamp 51814 Event: message-summary Content-Length: 0

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