May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dlz-ldap-enum - expose LDAP data to FreeSWITCH via ENUM
by Daniel Pocock 17 May '12

17 May '12

I've recently released a dlz ENUM module for the bind9 nameserver: http://www.opentelecoms.org/dlz-ldap-enum Basically, it handles ENUM queries from FreeSWITCH, Kamailio, Asterisk, repro, Lumicall, searches for the phone number in LDAP, and if found, returns the email address as both a SIP address and Jabber address This should make it even easier than ever before to get federated VoIP up and running using email addresses interchangeably with phone numbers. If the data already exists in LDAP as an address book, then just install bind9, install the module and you're up and running. Regards, Daniel

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Re: [Freeswitch-users] Loud whilte noise during sleep
by Phil Quesinberry 17 May '12

17 May '12

** Disclaimer: this is probably a really bad idea ** Just in case this is frustrating anyone else: Silence is generated by the switch_generate_sln_silence function(s) in switch_resample.c, 'divisor' is the variable which determines the noise volume - the higher the value, the lower the noise volume. So, a quick fix for this problem would be to hard-code a value like 1400 for 'divisor' within those functions, by adding divisor = 1400 before the for loop and recompiling. This will affect ALL silence generated by FreeSwitch so it would be better to find where this is set within the sleep function so it only affects noise generated there. Better yet, if there's a way to accomplish this without hacking the code, (i.e. a variable that can be set), then by all means do that instead (and let me know what it is). Setting send_silence_when_idle to the desired value curbs the noise during sleep but it sends silence during ringback too, at least in my configuration here. Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Portuguese Sounds
by Antonio Modesto 17 May '12

17 May '12

Hi, Does anybody knows if there is a portuguese version of the default freeswitch sounds? I only have found it in English and Russian.

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Fax tones for India ?
by Anita Hall 17 May '12

17 May '12

Hi In spandsp.conf.xml, the following tones are given for North America. And similarly, others are given for UK and Germany. What will be the Fax tones for India ? 20 <descriptor name="1"> 21 <tone name="CED_TONE"> 22 <element freq1="2100" freq2="0" min="500" max="0"/> 23 </tone> 24 <tone name="SIT"> 25 <element freq1="950" freq2="0" min="256" max="400"/> 26 <element freq1="1400" freq2="0" min="256" max="400"/> 27 <element freq1="1800" freq2="0" min="256" max="400"/> 28 </tone> 29 <tone name="REORDER_TONE"> 30 <element freq1="480" freq2="620" min="224" max="272"/> 31 <element freq1="0" freq2="0" min="224" max="272"/> 32 </tone> 33 <tone name="BUSY_TONE"> 34 <element freq1="480" freq2="620" min="464" max="516"/> 35 <element freq1="0" freq2="0" min="464" max="516"/> 36 </tone> 37 </descriptor> regards, Anita

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Re: [Freeswitch-users] mod_callcenter real answer
by Евгений 17 May '12

17 May '12

Hi, Gustavo. Thank you for the patch. For information - I think there is new version of mod_callcenter for you in GIT, because lines in your patch is on the new places in new mod_callcenter.c. Do you have issue with no ring_ready sound on client side?Best Regards, Evginey. http://blog.buchnev.ru ------------------------------------------------------------- Evginey: These are the changes we made: --- mod_callcenter.c.orig2012-01-11 20:23:34.000000000 -0300 +++ mod_callcenter.c2012-01-26 15:31:15.000000000 -0300 @@ -1420,6 +1420,7 @@ switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Name", h->member_cid_name); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Number", h->member_cid_number); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-DNIS", member_dnis); +switch_event_add_header(event, SWITCH_STACK_BOTTOM, "CC-Member-Joined-Time", "%" SWITCH_TIME_T_FMT, t_member_called); switch_event_fire(&event); } @@ -2408,7 +2409,7 @@ } /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ -switch_channel_answer(member_channel); +switch_channel_pre_answer(member_channel); /* Grab the start epoch of a channel */ times = switch_channel_get_timetable(member_channel); On May 15, 2012, at 7:19 PM, Gustavo Mársico wrote: Hi Evginey I made that change a few days ago, due a scenario where dialer is used. In fact, dialer dials to a queue first instead of customer. All I did was replace in the code "answer" by "pre_answer" in some place. I've no the code right now, but I can send it to you later. Regards Gustavo On May 15, 2012, at 6:15 AM, Евгений wrote: Hi, colleagues. Could I have real answer for client on mod_callcenter, when agent answers? Or there is another way to queue call between 2 FreeSwitch. For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? Best Regards, Evginey. http://blog.buchnev.ru

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SIP MESSAGE Source IP For Internal/External Config
by Jerry Richards 16 May '12

16 May '12

Hello, I'm using CentOS 5.7 and the internal sip_profile for eth0 and the external sip_profile for eth1. I can register phones on either network, make calls between either network, observe presence updates on either network. Everything is working well, except I have one-way Instant-Messaging (IM) between a softphone on the eth0 network and a softphone on the eth1 network. An IM from internal-to-external works fine. An IM from external-to-internal is not displayed at the softphone. In the latter case, I see Freeswitch sending the SIP MESSAGE with correct content, but it's source IP address is the eth1/external IP, instead of the eth0/internal IP. I'm not sure if this is a bug or a configuration issue? Do you know where Freeswitch pulls the source IP in this case? Thanks, Jerry

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2012 VoIP With Virtualization
by Chris Mylonas 16 May '12

16 May '12

Hi FS Users, What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA. If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... Thanks for your inputs Chris

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Invitation to connect on LinkedIn
by michael mendel via LinkedIn 16 May '12

16 May '12

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mod_callcenter real answer
by Евгений 16 May '12

16 May '12

Hi, colleagues. Could I have real answer for client on mod_callcenter, when agent answers? Or there is another way to queue call between 2 FreeSwitch. For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? Best Regards, Evginey. http://blog.buchnev.ru

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Loud white noise during sleep
by Phil Quesinberry 16 May '12

16 May '12

Is there a setting to reduce the white noise generated during sleep? Setting suppress-cng to true has no effect on that noise. Setting send_silence_when_idle to a non-zero value appears to disable ringback in early media, so that's not a good solution. Is there another variable that can be set to an appropriate value, i.e. "silence"? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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