- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Postgresql CDR
by Antonio Teixeira 21 Mar '13

21 Mar '13

Hello guys. I'm trying to implement CDrs in Postgresql , my objective is something like : http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv What i would like : Receiving The Entire CDR with all the channel variables , some are created inside the channel at 'runtime' so i need all the variables for processing. So does Postgresql in the core changes something to the above link? Is it possible to say in the schema something like give me all the variables and place it inside a specific field ? What do you guys recommend write a CDR into a dir and have an app to monitor and send all the files into a DB or use cdr_pg_csv. Thanks for the input Antonio

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Ringback tone
by Miha 21 Mar '13

21 Mar '13

Hi, I need one info regarding ringback tone. How ISP are sending for numbers which are unavalible, "number is temporary unavalible"? I guess this is send in ringback tone? but how to achive that with FS as I must bridge to avalible user and set action application ringback to set ringback application? I need to do this for unavalibe numbers, setting own ringbacktone, on the call should not be answerd as if the call is answered, it will be in billing. thanks! Miha

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New features needed when playing files - playback application
by Clive Lansink 21 Mar '13

21 Mar '13

There are two features I need in the playback application for playing sound files for an IVR application I am developing. Maybe one feature is there but I don't think the other is but perhaps people could advise me about that. 1. I need to be able to start at a specified time offset in the file. When playing is interrupted, there is a variable that can tell the number of milliseconds of audio that was played. If I want to resume playing, I need to be able to start from that point, or in fact any arbitrary point, in the file. I think I can achieve this with the uuid_fileman API command so maybe that will work. When I last looked at this I was confused as to whether the seek parameter was samples or milliseconds. 2. I want to be able to vary the speed of playback but in such a way as to keep the overall pitch of the recorded voice unchanged. There are algorithms that do this and I presume they work by removing samples from or adding samples into the audio stream. I think uuid_fileman also has a speed subcommand which kind of worked when I played with it, but it was not constant pitch. The application I am working on is a replacement of a system that delivers human narrated information to blind people over the phone. The existing system uses old analogue Dialogic cards and has these features, and the replacement system should also have these capabilities. Recorded bulletins of information can be quite lengthy, maybe an hour or more to read right through. So this is perhaps not typical of IVR systems. People need to be able to skip around and possibly speed up or slow down the reading, but without altering the pitch of the narrator's voice. I know I can create the necessary functionality if I have the above features in the playback application. Note that the application will also use TTS to turn textual information into speech but that is a separate issue and not relevant here. The application still needs to handle human narrated information. Can anyone comment on these features and, if something needs to be added, how easy it would be to do this and how it might be done? Thank you. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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External voicemail issue
by Elliot Farmer 21 Mar '13

21 Mar '13

Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258…, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help!

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How to add a SIP custom header when sending an SMS via socket interface
by Leonardo Pereira 21 Mar '13

21 Mar '13

Hi I'm using the Freeswitch to send SMSs via socket interface: bgapi chat sip|+13109927777|external/+5759988888(a)smsgw.com|test Is there a way to add a SIP custom header to the SIP MESSAGE request? Thanks in advance, Leonardo Nogueira de Sá Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77

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Java ESL MESSAGE event encoding
by Jimmy Chang 21 Mar '13

21 Mar '13

Hi, I'm using org.freeswitch.esl.client-0.9.3-SNAPSHOT library and I got message corrupt when I received message from SIP SIMPLE MESSAGE. The ?????? messages are chinese words. Here are the logs. Any suggestions would be appreicated. [2013/03/21 18:09:56] [DEBUG] read body line [Event-Name: MESSAGE] [2013/03/21 18:09:56] [DEBUG] read body line [Core-UUID: c275cf27-b718-48bb-a870-d48c236b27da] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-Hostname: asterisk-orig] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-Switchname: asterisk-orig] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-IPv4: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-IPv6: %3A%3A1] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-Local: 2013-03-21%2018%3A10%3A06] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-GMT: Thu,%2021%20Mar%202013%2010%3A10%3A06%20GMT] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-Timestamp: 1363860606137801] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-File: sofia_presence.c] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-Function: sofia_presence_handle_sip_i_message] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-Line-Number: 4502] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Sequence: 13794] [2013/03/21 18:09:56] [DEBUG] read body line [login: sip%3Amod_sofia%4010.0.110.33%3A5060] [2013/03/21 18:09:56] [DEBUG] read body line [proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [to_proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [from: 1018%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [from_user: 1018] [2013/03/21 18:09:56] [DEBUG] read body line [from_host: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [to_user: 1011] [2013/03/21 18:09:56] [DEBUG] read body line [to_host: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [from_sip_ip: 10.0.1.69] [2013/03/21 18:09:56] [DEBUG] read body line [from_sip_port: 5060] [2013/03/21 18:09:56] [DEBUG] read body line [to: 1011%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [subject: SIMPLE%20MESSAGE] [2013/03/21 18:09:56] [DEBUG] read body line [type: text/plain] [2013/03/21 18:09:56] [DEBUG] read body line [from_full: %3Csip%3A1018%4010.0.110.33%3E%3Btag%3D18420] [2013/03/21 18:09:56] [DEBUG] read body line [sip_profile: internal] [2013/03/21 18:09:56] [DEBUG] read body line [max_forwards: 70] [2013/03/21 18:09:56] [DEBUG] read body line [DP_MATCH: 1011%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [skip_global_process: true] [2013/03/21 18:09:56] [DEBUG] read body line [dest_proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [Nonblocking-Delivery: true] [2013/03/21 18:09:56] [DEBUG] read body line [Content-Length: 11] [2013/03/21 18:09:56] [DEBUG] read body line [] [2013/03/21 18:09:56] [DEBUG] read body line [Jimmy??????] Regards, Jimmy

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Missing audio sometimes
by jay prakash 21 Mar '13

21 Mar '13

Hi, I am using PRI card in freeswitch with freeTDM. Sometimes i am not getting audio in both inbound and outbound calling. Please help me to find out the issue. *Test Environment* Operating System :- Ubuntu 12.04 LTS Kernel version :- *3.2.0-23-generic* * *PCIe slot version :- *x1 PCI Express x1 * Freeswitch Git version :- 4319bc8bd9ac6be1391d67282d1bf0ab1fbeb3d6 is the commit id. You can simply use the git repository at git(a)git.freeswitch.org/freeswitch.git and install from it. Libpri version :- 1.4.13* * * *Dahdi version* :- 2.6.1* * * * * * * *Thanks & Regards* *JAY* * *

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Building Fs - Remake
by Antonio Teixeira 21 Mar '13

21 Mar '13

Hello guys. On a brand new Centos 6.3 Citrix Xen Server VM. Make fails with : http://pastebin.freeswitch.org/20700 Maybe should i try Centos 6.2 ?? Or other distro , I'm trying to find one that is stable with Freeswitch been more used in production and more battle tested. Thanks Antonio

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Poor audio quality due to rtp timing issues
by Garey Arrington 21 Mar '13

21 Mar '13

When making inbound and outbound calls I am getting slightly less than acceptable audio quality after a few seconds. The machine is a 64 bit Centos 5.5 vm running on Xen with kernel version: 2.6.18-194.el5 I am getting warnings like with each call: 2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME not supported, changing our end from 30 to 20 and 2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global timer resolution to 10ms to handle interval 30 I have tried the fix that I have consistently found for this issue which is to add this to external.xml: <param name="rtp-autofix-timing" value="false"/> But this makes the audio quality so bad you can hardly make out anything. Can someone point me in a better direction for a solution to this problem?

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FreeSWITCH-Portal with ember.js - Another GUI for FS fans
by Usama Zaidi 20 Mar '13

20 Mar '13

Hi, You might also want to look at metero framework, http://meteor.com/ -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, March 20, 2013 11:26 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 224 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..."

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