- FreeSWITCH-users - FreeSWITCH Mailing Lists

Custom Authentication
by Colin Morelli 14 Apr '16

14 Apr '16

Hey all, I'm looking for a way to support custom authentication with FreeSwitch. What I am hoping to get is similar to mod_xml_curl directory, but where FS will provide both user and password to my service. The use case I'm looking to achieve is to support token/key based authentication in addition to standard SIP auth. Is this something that's possible in FS? Any other suggestions? Thanks

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How to register to a gateway/provider using an outbound proxy but have realm value in request URI?
by Mervyn Yeo 14 Apr '16

14 Apr '16

Hi all, I'm trying to register with a provider without success. I'm able to do it from my snom 300 phone as well as zoiper softphone. On the phones, all I did was set <fake substitute values below> account : username registrar : sip.provider.com outbound proxy : 8.8.8.8 and it sends the register to 8.8.8.8 with a register packet like so in http://pastebin.com/XME4UknQ in fs, i created a provider with values http://pastebin.com/Q6i6Njxu and it's not registering. the difference in the sip trace is that although the request is sent to network address 8.8.8.8 the "Request-URI Host Part" uses the values from my proxy setting instead of "sip.provider.com". Sip trace from fs here -> http://pastebin.com/Pfgsu28s (Please ignore the user agent, this was sent from fs. I was experimenting with the "user-agent-string" param). How can I setup fs so that it'll send to the outbound proxy address but the Request-Line still shows REGISTER sip:sip.provider.com SIP/2.0 Thanks, Mervyn

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Re: [Freeswitch-users] Billsec in dialplan
by jelena＠misticnabica.hr 13 Apr '16

13 Apr '16

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module question
by Chris Mandra 13 Apr '16

13 Apr '16

Hey guys - I've been working on a ladspa module and I thought that after I make a change to it and recompile I can just run reload and hear my changes, but it's not working, so I tried unload and reload and that's not allowing me to hear my changes either. Restarting freeSWITCH does allow me to hear my changes. Any ideas why this would be happening. I'm running 1.6 I know about the how the modules shutdown function is supposed to work ( https://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Reloading_You…"N.B. your module's Shutdown function must accomplish a complete reversion of any setup that happens in it's Load function for a reload to work properly.") and this used to work without issue, so I'm wondering what might be going on here. Any suggestions appreciated, thanks, chris

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SMS/SMPP CDR
by Uday kumar 13 Apr '16

13 Apr '16

Hello All, I am using SMS/SMPP on Freeswitch 1.6 with CentOS 7.2. Is this possible to mange CDR report for SMS/SMPP in database like from user, to user, message and time save in database? Please advise. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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Thursday FreeSWITCH Bug Hunt - TOMORROW!
by Ítalo Rossi 13 Apr '16

13 Apr '16

FreeSWITCHers, If you have questions, need attention for any JIRA you reported or want to contribute, join us! Tomorrow 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees witch.org/vc/#/?autocall=888 "https://conference.freeswitch.org/vc/#/?autocall=888" ) Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc> What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

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Cannot get automatic call recording working.
by Piotr Starzyk 13 Apr '16

13 Apr '16

I’m running FreeSWITCH in console mode on a Windows 7 machine, using unmodified demo setup. I’m having a hard time getting automatic call recording to work. I’m using the following instructions to record a call (by modifying Local_Extension dialplan): https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+record_… *Record Calls To Extensions* To record all phone calls between extensions do the following. Make a directory under freeswitch/recordings/archive/. Then edit Local_Extension in dialplan in conf/dialplan/default.xml <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-8])$"> Then add the following actions. <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2011"/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=FreeSWITCH"/> <action application="set" data="RECORD_COMMENT=FreeSWITCH"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> But when I place a call, no files are getting created in the recordings or archive folder. I tried calling both ways, still nothing. I am running in admin mode, and have firewalls disabled. When looking through the logs, the only two lines relevant to the recording are the following: 892ef279-18c7-4d8d-8795-fa1c366b68c7 EXECUTE sofia/internal/1000(a)10.10.10.77 bind_meta_app(2 b s record_session::C:/Program Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav) 892ef279-18c7-4d8d-8795-fa1c366b68c7 2016-04-11 15:16:58.521344 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::C:/Program Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav >From FreeSWITCH cookbook, I found out they allow manual recording by pressing *2 on receiving end. When I tested that, it did work, and the wav file was created in the recordings directory, so the recording module seems to be working. But for some reason, I still can’t get the automatic session recording to work. What am I doing wrong? Thanks.

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Re: [Freeswitch-users] Call forced held after answer
by jelena＠misticnabica.hr 13 Apr '16

13 Apr '16

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Call forced held after answer
by Adrian Thompson 12 Apr '16

12 Apr '16

Hello, Any insight into this would be very helpful! I have exhausted my search of Freeswitch and Polycom. I have no idea how to solve this issue L I have a great working setup with VVX 300s 400s 500s and a IP 6000, however there is a strange issue going on with an IP 7000 I have. When calling to or from IP 7000 1. It rings fine with audio on both ends 2. Press answer on either side 3. Call is immediately "Held" on IP 7000 4. Green lights flash on IP 7000 as if the call was placed on hold by the other phone, but call is not on hold by other phone 5. No audio (probably because on hold) 6. End call normally NAT settings are correct - Formatted IP 7000 and reset all config - Reinstalled firmware - Disconnected from provisioning server - Created manual config from scratch on phone web interface - Removed all encryption IP 7000 logs during call: 003219.281|net |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2. 003219.341|net |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2. 003219.500|net |4|03|rtosNetwork: netwTask() - Can't find associated CCB. 003220.080|net |4|03|rtosNetwork net02: netwSend() - sendto() call failed. fd 1123346772 errno=130 003220.861|net |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2. 003221.600|net |4|03|rtosNetwork: netwTask() - Can't find associated CCB. 003223.600|net |4|03|rtosNetwork: netwTask() - Can't find associated CCB. Then press hang-up: 003227.600|net |4|03|rtosNetwork: netwTask() - Can't find associated CCB. 003227.985|net |2|03|NWIF: nw_setlocalhold() - setting local hold (1) 2. 003242.982|sip |5|03|Can not decode the packet - Setting local hold (0) should be fine and it works on all other phones I have. - Setting local hold (1) after pressing hang-up is weird but is happening after hang-up anyway. - I wonder why that send error is in the logs. it looks like an operating system error.. Maybe a driver error? Thanks in advance!! Adrian

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nibblebill do not terminate calls properly
by Tanguy 12 Apr '16

12 Apr '16

Hello I would like to use nibblebilling for fraud prevention on international and premium numbers ( national or emergency calls should never be blocked and should still working even if the billing database is unavailable ) Unfortunately that never hangup my calls using standard dial plan *nibblebill.conf.xml* <configuration name="nibblebill.conf" description="Nibble Billing"> <settings>   <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx user=xxxx password=xxxx"/>  <param name="db_table" value="accounts"/>  <param name="db_column_cash" value="cash"/>  <param name="db_column_account" value="id"/>  <param name="global_heartbeat" value="20"/>  <param name="lowbal_amt" value="-100"/> <param name="lowbal_action" value="play ding"/>  <param name="nobal_amt" value="0"/> <param name="nobal_action" value="hangup"/>  <param name="percall_max_amt" value="10"/> <param name="percall_action" value="hangup"/> </settings> </configuration> *outbound route* <extension name="international" > <condition field="destination_number" expression="^00(\d{6,20})$" > <action application="nibblebill" data="flush"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=0.1"/> <action application="bridge" data="sofia/gateway/gw_idt/$1" /> </condition> </extension> *My balance* My balance is already negative select * from accounts ; id | cash ----------------------------------+----------- company.voip.mydomain.com | -0.789471 *A sample call* My balance already negative, the call will be billed but never blocked Dialplan: sofia/internal/5003(a)company.voip.domain.com Action nibblebill(flush) Dialplan: sofia/internal/5003(a)company.voip.domain.com Action set(nibble_account=${accountcode}) Dialplan: sofia/internal/5003(a)company.voip.domain.com Action set(nibble_rate=0.1) EXECUTE sofia/internal/5003(a)company.voip.domain.com nibblebill(flush) EXECUTE sofia/internal/5003(a)company.voip.domain.com set(nibble_account=company.voip.domain.com) 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003(a)company.voip.domain.com SET [nibble_account]=[company.voip.domain.com] EXECUTE sofia/internal/5003(a)company.voip.domain.com set(nibble_rate=0.1) 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003(a)company.voip.domain.com SET [nibble_rate]=[0.1] 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute to account company.voip.domain.com 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new billing on 644488fc-ff04-11e5-9a27-fd2791153af9 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed since last bill time of 2016-04-10 12:10:09 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033 to company.voip.domain.com (Call: / 0.000000 so far) 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='company.voip.domain.com'] 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account company.voip.domain.com (balance = -0.860504) *Using b-leg only* Alternatively i tried an alternative dialplan ( Even if i don't relay understand what is the meaning of b-leg billing ) <action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1" /> The result is better because the pending call is hanged up when the balance reach 0. 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of -0.003237 fell below allowed amount of 0.000000! (Account company.voip.domain.com) But if i make a new call ( when my balance is negative ) , the caller party is immediately hanged but this did not cancel immediately the bridge: The called party ring and can stay bridged for 61 seconds after answer.

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