February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Polycom BLF w/ OpenSIPS and FreeSWITCH not working
by Vik Killa 14 Feb '12

14 Feb '12

I am unable to get BLF working when I have OpenSIPS in front of FreeSWITCH Would anybody know what is going wrong? 72.43.207.158 is OpenSIPS (test1.voip.demo.i-evolve.net) 72.43.207.155 is FS Here is a SIP trace take from FS on a REGISTER/NOTIFY/SUBSCRIBE: recv 725 bytes from udp/[72.43.207.158]:5060 at 13:30:15.670418: ------------------------------------------------------------------------ REGISTER sip:72.43.207.155:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKf54f.188e1555.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKf59d03a4266B7F41 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net> CSeq: 1 REGISTER Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 30 Expires: 3600 Content-Length: 0 Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 707 bytes to udp/[72.43.207.158]:5060 at 13:30:15.672807: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKf54f.188e1555.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKf59d03a4266B7F41 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net>;tag=B3eFSHXS8Q0vN Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 freeswitch@internal> freeswitch@internal> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="test1.voip.demo.i-evolve.net", nonce="079dfd86-5710-11e1-af11-2f1196b916eb", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 995 bytes from udp/[72.43.207.158]:5060 at 13:30:15.692185: ------------------------------------------------------------------------ REGISTER sip:72.43.207.155:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKc54f.4c7b9b9.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bK7a3b5bcD257C8B9 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net> CSeq: 2 REGISTER Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Authorization: Digest username="1000", realm="test1.voip.demo.i-evolve.net", nonce="079dfd86-5710-11e1-af11-2f1196b916eb", qop=auth, cnonce="CNyXC4dNuYJCV0G", nc=00000001, uri="sip:test1.voip.demo.i-evolve.net", response="4ff00198d12440fbc2aa14dcf9fb213b", algorithm=MD5 Max-Forwards: 30 Expires: 3600 Content-Length: 0 Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 711 bytes to udp/[72.43.207.158]:5060 at 13:30:15.713022: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKc54f.4c7b9b9.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bK7a3b5bcD257C8B9 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net>;tag=cc87tceX50pFH Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 CSeq: 2 REGISTER Contact: <sip:1000@72.43.207.130:1666>;expires=3600 Date: Tue, 14 Feb 2012 13:30:15 GMT User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> Content-Length: 0 ------------------------------------------------------------------------ send 964 bytes to udp/[72.43.207.158]:5060 at 13:30:15.827086: ------------------------------------------------------------------------ NOTIFY sip:1000@72.43.207.130:1666 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.155;rport;branch=z9hG4bK6Saj311cpBygK Route: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> Max-Forwards: 70 From: <sip:1000@test1.voip.demo.i-evolve.net>;tag=DN10v7y029c2c To: <sip:1000@test1.voip.demo.i-evolve.net> Call-ID: df13f894-d1b2-122f-6aa8-928cc183345f CSeq: 24277043 NOTIFY freeswitch@internal> freeswitch@internal> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer freeswitch@internal> freeswitch@internal> Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:1000@test1.voip.demo.i-evolve.net ------------------------------------------------------------------------ recv 543 bytes from udp/[72.43.207.158]:5060 at 13:30:15.847271: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 72.43.207.155;received=72.43.207.155;rport=5060;branch=z9hG4bK6Saj311cpBygK From: <sip:1000@test1.voip.demo.i-evolve.net>;tag=DN10v7y029c2c To: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=BEC991BF-465E06A8 CSeq: 24277043 NOTIFY Call-ID: df13f894-d1b2-122f-6aa8-928cc183345f Contact: <sip:1000@72.43.207.130:1666> Record-Route: <sip:72.43.207.158;lr;ftag=DN10v7y029c2c> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ freeswitch@internal> recv 796 bytes from udp/[72.43.207.158]:5060 at 13:30:16.653565: ------------------------------------------------------------------------ SUBSCRIBE sip:1004@72.43.207.155:5060 SIP/2.0 Record-Route: <sip:72.43.207.158;lr;ftag=D90A32FE-DB2A2443> Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bK9c9.5db2f931.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKe4c1b0d5A735EFD6 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 To: <sip:1004@test1.voip.demo.i-evolve.net> CSeq: 1 SUBSCRIBE Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 30 Expires: 3600 Content-Length: 0 X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 864 bytes to udp/[72.43.207.158]:5060 at 13:30:16.656893: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bK9c9.5db2f931.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKe4c1b0d5A735EFD6 Record-Route: <sip:72.43.207.158;lr;ftag=D90A32FE-DB2A2443> From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 To: <sip:1004@test1.voip.demo.i-evolve.net>;tag=P0syeQe6vUN3 Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 CSeq: 1 SUBSCRIBE Contact: <sip:1004@72.43.207.155:5060> Expires: 3600 User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ send 1300 bytes to udp/[72.43.207.130]:1666 at 13:30:16.731407: ------------------------------------------------------------------------ NOTIFY sip:1000@72.43.207.130:1666 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.155;rport;branch=z9hG4bK723a5vjgKmm3e Max-Forwards: 70 From: <sip:1004@test1.voip.demo.i-evolve.net>;tag=P0syeQe6vUN3 To: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 CSeq: 165317323 NOTIFY Contact: <sip:1004@72.43.207.155:5060;transport=udp> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: presence Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/xpidf+xml Content-Length: 460 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <status> <note>Registered</note> </status> <presentity uri="sip:1004@test1.voip.demo.i-evolve.net;method=SUBSCRIBE" /> <atom id="sip:1004@test1.voip.demo.i-evolve.net"> <address uri="sip:mod_sofia@72.43.207.155:5060;user=ip" priority="0.800000"> <status status="" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> -----------------------------------------------------------------------

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mod_callcenter overloading database when using MySQL via unixODBC
by Vik Killa 14 Feb '12

14 Feb '12

I have 2 FS servers sharing a mysql database via ODBC. There are 2 queues loaded in mod_callcenter and there is absolutely zero activity around mod_callcenter. The mysql.log is showing many queries a few times per second like: "select 1" "SELECT queue,uuid,session_uuid,cid_number,cid_name,joined_epoch,(1329162473-joined_epoch)+base_score+skill_score AS score, state, abandoned_epoch FROM members WHERE state = 'Waiting' OR state = 'Abandoned' OR (serving_agent = 'ring-all' AND state = 'Trying') ORDER BY score DESC" Even if i run "callcenter_config queue unload support1@default" to unload the queues from both FS servers, the mysql.log still doing the queries. The only way the queries stop is if i do "unload mod_callcenter" Here is my callcenter.conf: <configuration name="callcenter.conf" description="Call Center Module"> <settings> <param name="odbc-dsn" value="freeswitch:freeswitch:freeswitch"/> </settings> <queues> <queue name="support1@default"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="local_stream://moh/8000"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="max-wait-time-with-no-agent-time-reached" value="5"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> <queue name="support2@default"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="local_stream://moh/8000"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="max-wait-time-with-no-agent-time-reached" value="5"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> </queues> </configuration>

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Re: [Freeswitch-users] Rotate cdr-records as root user (when FS is running from freeswitch user).
by Henrik Aagaard Sørensen 14 Feb '12

14 Feb '12

I just tested the killall -HUB freeswitch as well, and the cdr records does not get rotated. This is my cdr_csv.conf.xml (the dots ... is just so it didn't take up the entire screen when copy-paste): <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="example"/>   <param name="rotate-on-hup" value="true"/>  <param name="legs" value="a"/>  <param name="master-file-only" value="true"/> </settings> <templates>  <template name="example">"${caller_id_name}","${caller_id_number}",...    </templates> </configuration> 2012/2/13 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com>: > Hi Vitalie. > > I've already set rotate-on-hub to true. But I prefer running the > fs_cli command as I do not want the log files to be rotated as well. > And if I set the rotate-on-hub to false in the log-conf, no files get > rotated on hub. Not even the cdr records. > > On Mon, Feb 13, 2012 at 3:05 AM, Vitalie Colosov <vetali100(a)gmail.com> wrote: >> Ensure the following line is uncommented in cdr_csv.conf.xml >> >> <param name=”rotate-on-hup” value=”true”/> >> >> >> To rotate, execute from commandline: >> >> killall -HUP freeswitch >> >> You can add this command to cron so it will be executed automatically. >> >> >> >> >> 2012/2/12 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> >>> >>> I'm having FS running from the user freeswitch on CentOS 6. >>> >>> When calling (as the user root): >>> /usr/local/freeswitch/bin/fs_cli -x 'cdr_csv rotate' >/dev/null >>> nothing happens. The cdr-records does not get rotated. >>> >>> Can anyone help me on why this is happening and how I can rotate the >>> files as root? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >>

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rad client
by Miha Zoubek 14 Feb '12

14 Feb '12

Hi, I need one information about radius behaviour regarding voip cdr (AAA). I know I get two stop packet and two start packets for every call that is made. For one softswitch I get session ID same for all paskets (stop&start for a leg, and stop&start for b leg), on other softswitch I get one session ID for one stop&start packet and other ID for second stop&start packet. So, for one softswitch as is only one ID I get one enetry in sql table, for other softswitch I get two enetries in my sql table as I get two session IDs. Which behaviour is right? Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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BLF and presence_map for conference and fifo?
by Tim St. Pierre 14 Feb '12

14 Feb '12

Hello, Just upgraded to FreeSWITCH Version 1.0.head (git-eeeaf41 2012-02-10 17-27-50 -0600) The BLF buttons on the phones now work much better, but I lost BLF status for conference rooms and parking slots. This worked just fine before, now it's broken. I did notices this new thing called presence_map.conf.xml, as well as a reference to it in the sip profile, but I can't find any documentation on how it works. What changed? My older version is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) (BLFs work for park and conference, but go out when a phone rejects a second call). Any thoughts? -Tim

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Displaying the caller id on each agent's cell phone when transferring calls to a queue using mod_callcenter
by Yungwei Chen 14 Feb '12

14 Feb '12

Hi, With the following channel variables, I am able to display caller ids (on SIP clients) for agents sitting behind SIP clients. origination_caller_id_name=test origination_caller_id_number=1234 But the same doesn't work if agents are using cell phones. It could be that my SIP provider doesn't support this. So I'm wondering if any of you got this working and please share the name of your SIP provider. Thanks.

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How to control Total Number of inbound and Outbound calls - through Gateway
by Lloyd Aloysius 13 Feb '12

13 Feb '12

Hi All, I want find a best way to achieve the following task. How to control Total Number of inbound and Outbound calls - through Gateway. 1. Domain abc.mydomain.com 2. How to control total number of calls through the gateway. But not between extensions. For Example for this domain I want to allow total 10 calls [ inbound calls + outbound calls] Thanks Lloyd

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FreeSWITCH ends the calls after receiving DTMF from the Callee.
by Srini K 13 Feb '12

13 Feb '12

Hi,I have configured FreeSWITCH to receive DTMF by <param name="liberal-dtmf" value="true"/> <variable direction="both" name="dtmf_type" value="info"/> in both sip_profile internal and external files. Iam using mod managed and I have subscribed to receive DTMF events.I can process DTMF from Caller without any problem. When I receive DTMF from the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH and immediately FreeSWITCH disconnects the call. Snapshot of the log is... [DEBUG] sofia.c:7229 INFO DTMF(1) [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy [BREAK] [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via DTMF [DEBUG] switch_ivr_bridge.c:384 Send signal sofia/external/xxxxxxxxxx(a)10.10.2.3 [BREAK] [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/yyyyyyyyyy] Whether Iam missing anything in the config? Thanks in advance.Srini

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Re: [Freeswitch-users] life360 feature with FreeSWITCH??
by Jerre Cope 13 Feb '12

13 Feb '12

I think you're on the wrong project. See OpenStreetmap.org and OpenLayers.org Geolocation/geocoding has nothing to do with SIP or FreeSWITCH. Tools and Wikis on the sites above will give you all the tools you need to build your own slippery map and place your points of interest via database. You will probably collect your GPS coordinates on a public website and and maintain your userbase and maps on a private server. You will need to do alot of coding, but the geospacial rocket science and cool mapping tricks are done with the code on those websites. Did you know you can get an opensource map of the whole world in 6 GB. OpenSource WOW! I'm not going to answer any more of your questions here. I'm just informing you that you need take this quest to the geniuses I've mentioned above, and not the FreeSWITCH geniuses.

1 0

Registration Count
by Yuriy Nasida 13 Feb '12

13 Feb '12

Hello guys! Is there the standart method to limit count of simultaneous registrations for some user? I wiki I have found only one similar ability http://wiki.freeswitch.org/wiki/API_sofia_count_reg But... It is for dialplan. I use curl module with php scripts for dynamic directory.So I can use this API from php script before sending the xml to FS. But is there more simple ways ? Btw, API 'sofia_count_reg' doesn't work with 'new' git'FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)' But! API 'sofia_count_reg' works with 'old' gitFreeSWITCH Version 1.0.head (git-54b4b08 2011-11-04 16-58-35 -0500) Please adviseThanks.

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