April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch 1.3.7 Final - mod_freetdm Compilation Failure
by Arpit Gadle 21 Apr '13

21 Apr '13

Hi All, I had downloaded *freeswitch-1.3.17-final.tar.gz *from http://git.freeswitch.org/git/freeswitch/. ./bootstrap.sh and ./configure steps were completed successfully but make command failed with following error message *CC ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:77: error: ‘SCT_MAX_NET_ADDRS’ undeclared here (not in a function)* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:105: error: ‘MW_MAX_NUM_OF_PEER’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:109: error: ‘MW_MAX_NUM_OF_INTF’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:112: error: ‘MW_MAX_NUM_OF_CLUSTER’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:129: error: expected declaration specifiers or ‘...’ before ‘SbMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:130: error: expected declaration specifiers or ‘...’ before ‘MwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:131: error: expected declaration specifiers or ‘...’ before ‘NwMgmt’* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:804: error: expected declaration specifiers or ‘...’ before ‘MwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:805: error: expected declaration specifiers or ‘...’ before ‘NwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:806: error: expected declaration specifiers or ‘...’ before ‘HiMngmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:807: error: expected declaration specifiers or ‘...’ before ‘SbMgmt’* *cc1: warnings being treated as errors* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c: In function ‘sngss7_save_iam’:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:3340: error: format ‘%d’ expects type ‘int’, but argument 10 has type ‘size_t’* *make[6]: *** [ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo] Error 1* *make[5]: *** [../libfreetdm.la] Error 2* *make[4]: *** [all] Error 1* *make[3]: *** [mod_freetdm-all] Error 1* *make[2]: *** [all-recursive] Error 1* *make[1]: *** [all-recursive] Error 1* *make: *** [all] Error 2* * * Any help is appreciated. Thanks & Regards, Arpit Gadle

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ftmod_zt.c error
by Ashish gautam 21 Apr '13

21 Apr '13

Hi, I am getting this error on the console. What could be the possible reasons for this and how to get rid of that: 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1350 [s1c16][1:16] Event 6 is not dmtf related. Skipping one media read cycle 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1352 [s1c16][1:16] Skipping one IO read cycle due to events pending in the driver queue 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) Thanks in advance. -- Ashish

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Respect the request URI domain and To URI domain when bridging
by José Luis Millán 21 Apr '13

21 Apr '13

Hi all, Every call bridged though a specific 'gateway' does rewrite the SIP request URI domain and To URI domain values to the gateway's 'proxy' parameter value. I would like to respect such values and receive in my proxy the SIP messages with their original request URI domain and To URI domain. This is, I would like to be able to send a SIP message with the following request URI and To headers, to the IP address 2.2.2.2 ``` INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:d@1.2.3.4>``` ``` Is it possible to handle this with FreeSwitch? I have followed the FS wiki but I've been unable to handle this. Thanks in advance. -- José Luis Millán

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Mod_xml_radius features
by Dmitry Erankevich 20 Apr '13

20 Apr '13

Dear friends, I’m researching the freeswitch’s deploying perspective in our current infrastructure. One of the case is integration with a billing system. I have tried various modules but cannot find dynamic registration feature. And maybe Mod_xml_radius is what I need but unfortunately cannot understand how it works. Could you please help me to find out if: 1. Mod_xml_radius provides sip dynamic registration with Radius protocol. No xml files need. User database can be located in Radius server side. 2. It is possible to mix somehow radius registration with regular xml files. 3. With Mod_xml_radius is possible to apply maximum concurrent calls for the user. Thank you very much for your support, Dmitry.

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Voicemail ends record after 30 seconds
by Pascal Taube 20 Apr '13

20 Apr '13

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Audio Lost after 1 min
by Sean Devoy 20 Apr '13

20 Apr '13

Hi All, I have a report by a customer at a site that we had considerable trouble with NAT issues. I thought everything was now stable, but it sounds like it is not. She reports that she is only telling me now, but for some time when they call someone and get no answer and get directed to voicemail (the latest was a Sprint number) the audio from the remote answering machine cuts off before then end of the message. She did point out the call timer on her phone continued normally. She swears this does not happen when talking to a live person! First, I don't think FS can even tell if the b-leg was answered by voice mail differently from a person. Please correct me if that is incorrect. So, I am proceeding on the assumption that calls that are answered take less than 60 seconds! 45 seconds of the reported call were waiting for Sprint to give up on the cell phone and route to voicemail. The log just shows the call being answered . then Normal Clearing. I tend to always suspect NAT or crappy Polycom 335s for this customer as that has been all my issues with them so far. Is there a NAT issues relating to max call length, etc? No other users (all cisco phones) on the same FS at other sites report this issue. Thanks in advance. Sean

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reloading mod_directory profiles
by Ira Tessler 20 Apr '13

20 Apr '13

Is there a way to reload the directory.conf (directory profiles) without restarting the server or doing a reload mod_directory? I use xml_curl to delivery the directory profiles and I need to by able to add new profiles on the fly. Thanks. Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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mod_dingaling sending wrong port numbers?
by Tom C 19 Apr '13

19 Apr '13

I'm trying to track down why Google Voice isn't working for me. When it stopped working a couple months ago, I learned that I needed to disable the video codec in autoload_configs/dingaling.conf.xml. That fixed things for a while, but for the past month it has been failing again. In my debug log for an INCOMING call, I see this: 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate mysubdomain.dyndns.com:17554 [27Pe5xE7PiNurvTC] 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate mysubdomain.dyndns.com:17555 [mnwhL1eq9SKrduUQ] 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates The domain is correct, but where is mod_dingaling getting those random port numbers? Those ports are not available on my network. Shouldn't mod_dingaling use the port range that is specified in autoload_configs/switch.conf.xml? How do I give mod_dingaling a range of ports to use? To put this in context, I'm trying to track down why Google Voice is no longer working for me. I got the "Accepted 0 of 0 candidates" errors a few months ago, but I removed the video codec from autoload_configs/dingaling.conf.xml, and the problem went away. Now those error messages are back, and I can't send or receive calls via Google Voice. Updated to the latest git on April 15th. (1.5.1b+git~20130415T194750Z~be248879ae (git be24887 2013-04-15 19:47:50Z)), and still the same problem as on the February build that I was running before.

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Jira Open Tickets and Patches (IMPORTANT INFORMATION)
by Ken Rice 19 Apr '13

19 Apr '13

For the last several weeks Ray and myself have embarked on a weekly meeting to go thru the outstanding tickets on Jira and work toward resolution on them. We have asked for reporters to provide updated information on many of them and a good number of those reporters have responded. To those that have updated their tickets as requested Thank You! To those ignoring the the requests for more information, if you have not responded to requests for more information within 14 days, your ticket will likely be closed unless you respond. This policy will be enforced on a ticket by ticket basis. With the core team making the final call. At some point in the future we will likely automate this process. Now, that being said, the goal here is not to just reduce the number of open tickets, but to make sure we are focusing efforts where we can actually do something about it. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Friday Free For All Activate
by Ken Rice 19 Apr '13

19 Apr '13

Join us for the Free For All, the Bridge is open! sip:888@conference.freeswitch.org Or use any of the DIDs K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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