- FreeSWITCH-users - FreeSWITCH Mailing Lists

Anyone used Instant messaging with FS
by Farrukh Ali 27 May '13

27 May '13

Dear all, I am trying to configure and use instant messaging with FS, can anyone give me a hint about configuration in FS and the soft phones? Regards, Muhammad Farrukh

2 1

Freetdm PRI Span Failover
by Ashish gautam 27 May '13

27 May '13

Hi, I have been using single span PRI card with FreeTDM but now I plan to use 8 Span card. I want to know if the channels on a particular span are all busy(or unavailable) at a time and a call needs to be originated, then how to automatically pass it through the other free span (failover kind of a thing) on which the channels are available? Please help! -- Ashish Gautam

1 0

missing wav files from yum install?
by Adam Cooley 27 May '13

27 May '13

Got freeswitch installed, and debugs on fs_cli show were missing audio files with a fresh install of 1.2.9 from yum on centos. 2013-05-25 19:48:27.071848 [ERR] mod_sndfile.c:202 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/ivr/ivr-to_call_the_freeswitch_conference.wav] [System error : No such file or directory.] Where do we get the default system wav files from? -a

1 0

SRTP media handling survey behaviour.
by Daniel Ivanov 26 May '13

26 May '13

I wanted to implement a mixed mode of srtp and zrtp crypto capabilities across our network and i was facing several issues. But succeeded nonetheless. What kicked me in the balls hardest was rfc3713(taken from memory) compliance of the UAs. Whenever i wanted to set SRTP to optional on the devices sent both avp and savp lines along with the according keys in the a=crypto lines in the sdp and the zrtp-hash too. If for a reason i wanted to force ZRTP on the channel, i was supposed to set sip_secure_media to false and that would disable srtp on the b-leg and zrtp handshake would take place in p2p mode. I was using csipsimple(pjsip) though and fs was just stripping the savp from the sdp and still passing the a=crypto lines. It turned out greedy and still started talking srtp instead of zrtp. I patched it myself to strip the a=crypto too if sip_secure_media is false and it worked like a german. I wanted to know if anyone has observed how other UAs handle the srtp negotiation and if it's worth proposing my patch as an improvement on the JIRA.

1 0

Strange issue with late negotiation
by Flavio Goncalves 25 May '13

25 May '13

Hello, I'm having a really strange issue on billing related to late negotiation. The call end up hanged with no BYE. There is a dialer sending thousands of calls through and OpenSIPS to a termination based on FreeSwitch. I have a SIP call flow as below. Some times during the day, when the volume is high, the UAC drops some 200Ok and FS send a reinvite in the middle of the Initial transaction. ----INVITE --------------- Proxy --INVITE ---------------> FS <----200 OK -------------- Proxy <---200 OK- ------------- FS <--REINVITE--------------- Proxy <-REINVITE--------------- FS ---481 leg does not exit-> Proxy ---481------------------> FS <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS ----CANCEL ---------------> Proxy ---CANCEL --------------> FS <---200 Ok ---------------- Proxy <----200 Ok ------------- FS FS is sending a reinvite before ACK comes from the client. I have two questions: 1) Is it valid to send a reinvite in the middle of an existing transaction? According to the RFC3261 Section 14. Note that a UAC MUST NOT initiate a new INVITE transaction within a dialog while another INVITE transaction is in progress in either direction. 1. If there is an ongoing INVITE client transaction, the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. 2. If there is an ongoing INVITE server transaction, the TU MUST wait until the transaction reaches the confirmed or terminated state before initiating the new INVITE. But it is confusing because just below it says the opposite. However, a UA MAY initiate a regular transaction while an INVITE transaction is in progress. A UA MAY also initiate an INVITE transaction while a regular transaction is in progress. 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK? If a UAS generates a 2xx response and never receives an ACK, it SHOULD generate a BYE to terminate the dialog. Best regards, Flavio E. Goncalves

4 5

How fs handle the sip-T message?
by xiaofengcanyuexp＠163.com 25 May '13

25 May '13

Hi, support team I know FS can pass out SS7 info via properity header. While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it? Thanks Windy ------------------- 2013-05-25

3 3

Assertion when Transcoding
by Guillermo Ruiz Camauer 24 May '13

24 May '13

I am getting the following assertion when attempting to record a G.729 call: freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. The system has a Sangoma D100 card installed. The call is established using the G.729 codec, and after playing some prompts, I am attempting to record an answer: freeswitch@192.168.9.151:5000@internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,GSM,mod_spandsp codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Sangoma G729,mod_sangoma_codec 18 total. . . . 2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File [/mnt/TEMP/beep-7.G729] 8000hz 2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY] freeswitch@192.168.9.151:5000@internal> freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. FreeSwitch dies after this. Note that this does not happen if I restrict everything to G711 (ULAW). Any thoughts? -- Guillermo Ruiz Camauer

2 2

Freeswitch versioning / tagging /releasemanagement
by mbo 24 May '13

24 May '13

On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I can see that there are many tags with a higher version, but older tag date. v1.3.0 1.3.0 release v1.3.1 Tagging version 1.3.1 v1.3.10 tag v1.3.10 v1.3.11 tag v1.3.11 v1.3.12 Retag v1.3.12 v1.3.13 tag v1.3.13 v1.3.14 tag v1.3.14 v1.3.15 tag v1.3.15 v1.3.16 tag v1.3.16 v1.3.17-final tag v1.3.17-final v1.3.2 Tagging version 1.3.2 v1.3.3 Tagging 1.3.3 v1.3.4 release v1.3.4 v1.3.5 release v1.3.5 v1.3.6 tag v1.3.6 v1.3.7 tag v1.3.7 v1.3.8 tag v1.3.8 v1.3.9 tag v1.3.9 v1.5.0 tag v1.5.0 What about those tags? How is the release management organized? Thanks Markus

4 3

Re: [Freeswitch-users] mod_cdr_csv is not working!
by Shahzad Bhatti 24 May '13

24 May '13

i open a jira and here is a link of that http://jira.freeswitch.org/browse/FS-5468 Shahzad Bhatti On Wed, May 22, 2013 at 8:50 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: mod_cdr_csv is not working! (Michael Collins) > 2. Re: One Way Audio (Michael Collins) > 3. Re: spam>spam>Re: One Way Audio (Moishe Grunstein) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 22 May 2013 08:39:26 -0700 > Subject: Re: [Freeswitch-users] mod_cdr_csv is not working! > Can you reproduce this on latest HEAD? If so, gather details and open a > Jira. > > -MC > > > On Wed, May 22, 2013 at 8:00 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com>wrote: > >> Hi, >> >> after loading *mod_cdr_csv, *when i originate test calls the cdr is not >> appended in Master.csv file most of the time but sometime cdr is appended >> in the Master.csv that show that mod_cdr_csv is configured. Also when i try >> to use *cdr_csv rotate* command on fs_cli it also not work. even i also >> try to do using the perl script example available on following url >> >> >> https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_in… >> >> i want to know why this is happening on my server and how i can fix these >> issues >> >> 1. cdrs are not updated in the csv file; >> 2. rotate command is not working; >> >> >> any reply is highly appreciated; >> >> Regards >> >> Shahzad Bhatti >> >>

1 0

Best performing codec on mobile/slow networks
by Daniel Ivanov 24 May '13

24 May '13

I've officially given up on AMR, due to it's licensing issues and sketchy supports for my mobile clients. I am looking for new options now as CsipSimple has a new codec pack supporting G.726.1, opus and codec2. Which do you think would perform best on mobile networks? And which has FS support for transcoding as well?

3 3