March 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Sending RTP data without SIP
by Georgi Stefanov 28 Mar '13

28 Mar '13

Hello All, Recently, I am having fun with freeswitch. I need to implement a module, which sends RTP data to previously specified IP and PORT. I do not want to have SIP/SDP session. Which is the best or just good enough way to do that? I understand my module should act as an endpoint then I need probably would need to create the session object, set the needed codecs, load (for example) the desired file for playing and start sending RTP. I had some debugging actions on freeswitch and found that functions sofia_glue_negotiate_sdp, sofia_glue_tech_choose_port are dealing with local and remote site of RTP connections. Could you advice me ?

2 1

[Freeswitch-dev] FreeSWITCH Community Conference Call Today
by Michael Collins 27 Mar '13

27 Mar '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27 We have a few quick items to discuss and then we will have Mark Crane give us an update on FusionPBX. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

[Freeswitch-dev] mod_fifo
by Thierry Panthier 26 Mar '13

26 Mar '13

Hi, I'm currently using a FreeSWITCH version from the stable branch (checked out on 29/01/2013), which works perfectly fine. Yesterday I tried to use version 1.2.7 (with the exact same configuration) but my fifo stopped working. Basically every call never leaves the queue, they hang in there *waiting* forever. I compared these versions and I noticed the following modification: - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*); + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*); What was the motivation for this change? If I revert the changes (use the old mod_fifo) everything works fine! Is anyone out there having a similar problem? Regards, Thierry Panthier

4 6

Re: [Freeswitch-dev] Unexpected call drop on PSTN network
by Ashish gautam 26 Mar '13

26 Mar '13

Hi, The issue got resolved. I did not set "ignore_early_media=true" in the dial string. Everything is fine after that. Thanks! On Tue, Mar 26, 2013 at 5:04 AM, < freeswitch-dev-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > > Today's Topics: > > 1. Re: sofia_glue - sdp with m=image 0 (Anthony Minessale) > 2. Unexpected call drop on PSTN network (Ashish gautam) > 3. FreeSWITCH Weekly News and Notes (Michael Collins) > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Date: Fri, 22 Mar 2013 11:54:48 -0500 > Subject: Re: [Freeswitch-dev] sofia_glue - sdp with m=image 0 > The log would have helped explain that you are using proxy_media mode > which is an important fact. > > Please try latest HEAD. And from now on report issues like this to > http://jira.freeswitch.org > > Bugs should not be fixed via the mailing list. > > > > > On Fri, Mar 22, 2013 at 2:01 AM, Wolfgang Pichler <wpichler(a)yosd.at>wrote: > >> sorry - i do have already reconfigured the audiocodes gateway which was >> sending this - and it is already in production... >> >> But i think the whole thing is also visible without trace, it is more a >> basic implementation question. >> >> The question is - is it really a good idea to overwrite the audio port - >> when there is also a image port in the sdp ? >> >> Here is the function i mean: >> >> >> >> switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t >> *tech_pvt, const char *sdp_str) >> { >> const char *err; >> char rip[RA_PTR_LEN] = ""; >> char rp[RA_PTR_LEN] = ""; >> char rvp[RA_PTR_LEN] = ""; >> char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL, >> *pe; >> int x; >> const char *val; >> switch_status_t status = SWITCH_STATUS_FALSE; >> >> if (zstr(sdp_str)) { >> sdp_str = tech_pvt->remote_sdp_str; >> } >> >> if (zstr(sdp_str)) { >> goto end; >> } >> >> if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p = >> (char *) switch_stristr("c=IN IP6 ", sdp_str))) { >> ip_ptr = p + 9; >> } >> >> if ((p = (char *) switch_stristr("m=audio ", sdp_str))) { >> port_ptr = p + 8; >> } >> >> if ((p = (char *) switch_stristr("m=image ", sdp_str))) { >> port_ptr = p + 8; >> } >> >> if ((p = (char *) switch_stristr("m=video ", sdp_str))) { >> vid_port_ptr = p + 8; >> } >> >> >> As you can see - it first checks for audio port - writes it to port_ptr, >> then checks for the image port - and if it does exists (also if it exists - >> but is zero - which is legal) - it does overwrite the audio port. >> >> So calls with sdp >> >> m=audio 12123 >> and >> m=image 0 >> >> will get dropped by freeswitch ! >> >> Thanks for you attention... >> >> br, >> Wolfgang >> >> 2013/3/22 Anthony Minessale <anthony.minessale(a)gmail.com> >> >>> post a trace with "sofia global siptrace on" >>> its too hard to diagnose without the exact trace. >>> >>> >>> On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler(a)yosd.at>wrote: >>> >>>> Hi all, >>>> >>>> i have encountered a strange behaviour - and i think i have found a >>>> possible problem within sofia_glue >>>> >>>> Following scenario: >>>> >>>> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer -> >>>> asterisk does forward offer -> ATA >>>> >>>> In this scenario it happend to me - that the T38 Offer (with corrent >>>> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk >>>> did forwarded without the T38 Offer (don't know why) - so also without the >>>> m=image sdp part - the ATA did answered correctly (without m=image part) -> >>>> asterisk did created a new sdp with m=image 0, without T38 parts - and >>>> returned this to freeswitch. >>>> >>>> Freeswitch did hangup the call - because sofia_glue did extracted the >>>> port 0 as audio port - and this is not a legal port... >>>> >>>> Sending port 0 is according to rfc legal - so freeswitch should ignore >>>> it in this case. >>>> >>>> the question is - if both values are given - which one is the value we >>>> need ? I think the audio port is the port which is of higher priority. >>>> >>>> So the workflow should be: >>>> - Try to extract audio port from m=audio part. >>>> - If you got it - and it is a valid port then go on >>>> - else try to extract audio port from m=image part... >>>> >>>> The function in question is sofia_glue_tech_proxy_remote_addr - in >>>> sofia_glue.c >>>> >>>> Is it generaly a good idea to use the m=image part as audio port ? Why >>>> is there no direct image port member in the pvt structure as with the video >>>> port ? >>>> >>>> Hope someone can help here... >>>> >>>> br, >>>> Wolfgang >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)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 >>>> >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale@hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888@conference.freeswitch.org >>> googletalk:conf+888@conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)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 >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)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 >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Ashish gautam <ashish(a)nms.co.in> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Date: Fri, 22 Mar 2013 16:23:43 +0530 > Subject: [Freeswitch-dev] Unexpected call drop on PSTN network > Hi, > > I am facing a strange issue with making outgoing calls to PSTN netwrok > through freetdm. The calls are successfully made to some Telecom networks > while they just drop abruptly and hangup on other networks. The calls are > being generated through inbound Even Socket originate action and put in a > public context which executes a perl script in the dialplan. > > Please help me out. Thanks in advance. > > -- > Ashish > > > ---------- Forwarded message ---------- > From: Michael Collins <marketing(a)cluecon.com> > To: freeswitch-users(a)lists.freeswitch.org, > freeswitch-dev(a)lists.freeswitch.org, > freeswitch-cluecon(a)lists.freeswitch.org > Cc: > Date: Mon, 25 Mar 2013 16:34:03 -0700 > Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes > Greetings! > > We are back after a brief hiatus. Our friend and colleague Brian West took > some much-deserved time off last week so the rest of us were a bit busier > than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel, > and ClueCon, so we're definitely glad he's back to work! > > On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked about Debian packaging with a little VoIP security and ZRTP > thrown in for good measure. Interestingly, a number of people stayed on the > call until 6pm EST! You are welcome to stay in the public conference room > for as long as you like. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark Crane scheduled to give us an update on FusionPBX. It's been > about 12 months since we last heard from him. We are looking forward to > seeing the improvements that have been added in the past year. > > We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save > the date: August 6-8, 2013. If you have any questions about being a > speaker, sponsor, or attendee then by all means contact us at this email > address. Remember that WebRTC is a big topic in the news right now and we > are looking for folks to talk about WebRTC in their open source telephony > projects and solutions. > > Have a great week and we'll talk to you again in April! > > -- > Michael S Collins > ClueCon Team > http://www.cluecon.com > 877-7-4ACLUE > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

1 0

[Freeswitch-dev] sofia_glue - sdp with m=image 0
by Wolfgang Pichler 26 Mar '13

26 Mar '13

Hi all, i have encountered a strange behaviour - and i think i have found a possible problem within sofia_glue Following scenario: Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer -> asterisk does forward offer -> ATA In this scenario it happend to me - that the T38 Offer (with corrent m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk did forwarded without the T38 Offer (don't know why) - so also without the m=image sdp part - the ATA did answered correctly (without m=image part) -> asterisk did created a new sdp with m=image 0, without T38 parts - and returned this to freeswitch. Freeswitch did hangup the call - because sofia_glue did extracted the port 0 as audio port - and this is not a legal port... Sending port 0 is according to rfc legal - so freeswitch should ignore it in this case. the question is - if both values are given - which one is the value we need ? I think the audio port is the port which is of higher priority. So the workflow should be: - Try to extract audio port from m=audio part. - If you got it - and it is a valid port then go on - else try to extract audio port from m=image part... The function in question is sofia_glue_tech_proxy_remote_addr - in sofia_glue.c Is it generaly a good idea to use the m=image part as audio port ? Why is there no direct image port member in the pvt structure as with the video port ? Hope someone can help here... br, Wolfgang

2 4

[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 25 Mar '13

25 Mar '13

Greetings! We are back after a brief hiatus. Our friend and colleague Brian West took some much-deserved time off last week so the rest of us were a bit busier than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel, and ClueCon, so we're definitely glad he's back to work! On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked about Debian packaging with a little VoIP security and ZRTP thrown in for good measure. Interestingly, a number of people stayed on the call until 6pm EST! You are welcome to stay in the public conference room for as long as you like. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark Crane scheduled to give us an update on FusionPBX. It's been about 12 months since we last heard from him. We are looking forward to seeing the improvements that have been added in the past year. We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save the date: August 6-8, 2013. If you have any questions about being a speaker, sponsor, or attendee then by all means contact us at this email address. Remember that WebRTC is a big topic in the news right now and we are looking for folks to talk about WebRTC in their open source telephony projects and solutions. Have a great week and we'll talk to you again in April! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

[Freeswitch-dev] Unexpected call drop on PSTN network
by Ashish gautam 22 Mar '13

22 Mar '13

Hi, I am facing a strange issue with making outgoing calls to PSTN netwrok through freetdm. The calls are successfully made to some Telecom networks while they just drop abruptly and hangup on other networks. The calls are being generated through inbound Even Socket originate action and put in a public context which executes a perl script in the dialplan. Please help me out. Thanks in advance. -- Ashish

1 0

[Freeswitch-dev] Freeswitch's capability of switching RTP transmission to new destination's address without any SIP signaling update
by Jyotshna Cherukuri 21 Mar '13

21 Mar '13

Hi, I am wondering if current version of FS has the capability to dynamically switch RTP trasnmission to the source address of RTP packets if the source address of RTP packets gets changed automatically while a call is already in progress . This might happen when a mobile handset gets a different IP on a access point reconnect during an ongoing call. Any help is greatly appreciated Thanks in advance Regards Jyotshna

3 2

[Freeswitch-dev] freetdm channel restarting error
by Ashish gautam 21 Mar '13

21 Mar '13

Hi, I have set up and installed a PRI line on my freeswitch machine. I am getting these warnings and errors when I start FS: 2013-03-21 10:38:11.294272 [WARNING] ftdm_io.c:3018 [s1c3][1:3] Channel not opened, proceeding anyway 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c4][1:4] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c5][1:5] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c6][1:6] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c7][1:7] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c8][1:8] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c9][1:9] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c10][1:10] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c11][1:11] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c12][1:12] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c13][1:13] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c14][1:14] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c15][1:15] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c17][1:17] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c18][1:18] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c19][1:19] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c20][1:20] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c21][1:21] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c22][1:22] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c23][1:23] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c24][1:24] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c25][1:25] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c26][1:26] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c27][1:27] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c28][1:28] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c29][1:29] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c30][1:30] -- T316 timed out, resending RESTART request 2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c31][1:31] -- T316 timed out, resending RESTART request 2013-03-21 10:38:41.314272 [NOTICE] ftmod_libpri.c:2159 [s1c4][1:4] -- Restart of channel completed 2013-03-21 10:38:41.314272 [WARNING] ftdm_io.c:3018 [s1c4][1:4] Channel not opened, proceeding anyway 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c5][1:5] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c6][1:6] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c7][1:7] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c8][1:8] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c9][1:9] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c10][1:10] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c11][1:11] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c12][1:12] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c13][1:13] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c14][1:14] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c15][1:15] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c17][1:17] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c18][1:18] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c19][1:19] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c20][1:20] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c21][1:21] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c22][1:22] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c23][1:23] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c24][1:24] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c25][1:25] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c26][1:26] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c27][1:27] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c28][1:28] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c29][1:29] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended Please help me to get out of this.

1 0

[Freeswitch-dev] 'make current' doesn't clean libs dirs
by Nathan Neulinger 20 Mar '13

20 Mar '13

Noticed this while trying to test the fax support and found that mod_spandsp wouldn't load due to functions not being defined. It looks like the 'make clean' and 'make modwipe' leave quite a few .o files around in the libs dirs. I'd imagine that most of the time, this has no impact, but when something in there changes...... -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

2 3