May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sdp manipulation help
by Daniel Ivanov 16 May '13

16 May '13

I am having trouble putting srtp to optional mode. UAs send sdp with rtp/avp and a:crypto lines which im permitting. But i want to decide whether the call uses srtp or zrtp. The UAs send also zrtp-hash. I export sip_secure_media to false but am still seeing the a:crypto lines being sent to the b-leg. I cut them off in my dialplan with set switch_r_sdp, but that breaks the whole negotiation, because the sdp isn't properly patched afterwards for nat( i also set a bunch of other vars afterwards). Is there a polite way of telling fs to remove the a:crypto lines and break the srtp and thus activating zrtp. Running 1.2.8git with inbound_zrtp_passthru to true.

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mod_cdr_csv is not add cdr data in csv file
by Shahzad Bhatti 16 May '13

16 May '13

Hi everyone, i am using *FreeSWITCH Version 1.5.1b+git~20130515T182820Z~37d109b107 (git 37d109b 2013-05-15 18:28:20Z)* on CentOS and for the mod_cdr_csv my conf. is as: <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="example"/>  <param name="debug" value="false"/> <param name="rotate-on-hup" value="false"/>  <param name="legs" value="a"/>  <param name="master-file-only" value="true"/> </settings> <templates> <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template> <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template> <template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template> <template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template> <template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template> <template name="opencdrrate">"${uuid}","${signal_bond}","${direction}","${ani}","${destination_number}","${answer_stamp}","${end_stamp}","${billsec}","${accountcode}","${userfield}","${network_addr}","${regex('${original_caller_id_name}'|^.)}","${sip_gateway_name}"</template> </templates> </configuration> but the issue is that when i use *cdr_csv rotate* on fs_cli then sometime it not works and also the cdr data is not added in the csv file sometimes any suggestions is highly appreciated Regards Shahzad Bhatti

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Re: [Freeswitch-users] IP change detected -> no internal profile
by info 16 May '13

16 May '13

On Tue, May 14, 2013 at 10:31 AM, Siri MM < sirimmfs(a)gmail.com > wrote: Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks! Hi Siri, I am a bit mystified as to how things are supposed to work if (how?) your IP get's switched between static and 127.0.0.1 (again, how?) This feels just wrong on so many levels :-) 127.0.0.1 normally is bound to the loopback interface, which, on linux, is called lo your other IP should be bound to an eth0 interface, or an internal ip and and the static IP on a router with some sort of NAT or port redirection. AFAIK linux should not allow you bind 127.0.0.1 to eth0 if it is allready bound to lo, so my best guess would be that you loose ethX and you are binding FS to all available interfaces... Alas, I don't know how to answer, because even if there should be a problem bringing up the internal profile on such a "switch", my guess is that that's probably *not* the problem you should be looking to fix for right now. Especially since, if somehow the IP really get's switched to 127.0.0.1, you will be talking only to yourself and I do not see how this should create a working setup (unless you use only an internal sip/(pri|bri) card or some USB channelbanks/gprs sticks, etc. and you only will be doing robocalls or automated IVR's ... Well, sorry mate that I can't help you more, but this really is a strange szenario, best regards, Ray

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IP change detected -> no internal profile
by Siri MM 16 May '13

16 May '13

Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks!

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Late mother's day gift - Advanced Do Not Call service!
by Sean Devoy 15 May '13

15 May '13

HI all, My mother has made a request and Mom's always get our best effort right? For those of you outside the US, we have a national Do Not Call Registry: www.donotcall.gov. You register your number there and in 30 days or less you are on the list. Telemarketers are required by law to check the list before calling and not call if you are on it. It does certainly help, but overseas call centers tend to ignore it and of course the excluded groups from the law "charities, political organizations, and telephone surveyors" may still call you. So it is simple enough for her to tell me the number that called and I add it to my dial plan that plays a recording that politely says "go pound sand." So help me take this to the next level: 1. What should I be returning to the caller's switch (like temp unavail) instead of a recording? How do I do that? 2. She has an ATA and analog phone (so extra features are limited), but I would love to be able to give her some kind of option like "hang up and dial *666 to add them to the list of band numbers" or maybe "hook-flash dial *666 and hang up" to xfer them to the dial plan that does that? All ideas are welcome and certainly appreciated. Thanks, Sean

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UUID in MESSAGE event
by Leonardo Pereira 15 May '13

15 May '13

Hi. I'm using the 'chat' command to send SMS: bgapi chat sip|+13229944553|external/+551991221234(a)MY.GW.COM|test message Is there a way to have some kind of UUID parameter in MESSAGE event? Leonardo Nogueira de Sá Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77

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Decline
by Blake Priddy 15 May '13

15 May '13

Ok I talked to you guys previously about this and our gateway told me I am sending the call to them incorrectly.. Which it is just 2 phones that are doing this. The rest are fine? http://pastebin.com/3ppLaatg Any thing in there throw a flag? -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Ken's News and Notes for FreeSWITCH....
by Ken Rice 15 May '13

15 May '13

Hey Guys, Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 and I said packages were coming. We now have both YUM and Debian Repo¹s for you guys to install from... For Debian check out http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packa ges For Centos check out http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_In stallation These will get you the stable branch for now... TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb¹s and rpm¹s for it -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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channel selection during outbound dialing
by mbo 15 May '13

15 May '13

We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in one group. In Asterisk it is possible to specify the channel selection mode by a prefix in the dial command (Dial(Zap/g2…) - ascending sequential hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2…) - descending sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I use a dial string like: freetdm/mygroup/a/ to dial out via a group. I'm not sure for what the "a" stand for. Are there also options for the channel selection mode available like in asterisk? Thanks Markus

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hangup extension
by Ashish gautam 15 May '13

15 May '13

Hi, Is there anything like hangup extension which gets executed after the call is hung up for which the UUID is same and based on that I can do things for the same call like we have in Asterisk? Thanks. Regards, --Ashish

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