May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Trouble With 'transfer' DP and Async Event Socket
by Joe Crivello [VT] 03 May '10

03 May '10

Hello list, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

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Trouble With 'transfer' DP and Async Event Socket
by Joe Crivello [VT] 03 May '10

03 May '10

Hello list, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

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nat problem
by babak yakhchali 03 May '10

03 May '10

Hi I got two 7941 ip phones connected to freeswitch (ip phone)192.168.22.57 -> 192.168.22.1 -> 192.168.11.30(fs) they can register fine but I can not call from ip phone to ip phone. when I use 3cx softphone in the same conditions I can call from ip phone to 3cx but not from 3cx to ip phone this is result of "sofia status profile internal" and when I get pcap on fs when invite is sent to ip phones nothing comes back Call-ID: 001bd432-a95c0003-942f6b14-17330728(a)192.168.22.56 User: 1005(a)192.168.11.30 Contact: "user" <sip:1005@192.168.22.56:5060 ;transport=udp;fs_nat=yes;fs_ path=sip%3A1005%40192.168.22.56%3A49551%3Btransport%3Dudp> Agent: Cisco-CP7941G/8.5.2 Status: Registered(UDP-NAT)(unknown) EXP(2010-05-02 14:11:11) Host: s-efrn2brk8zpcy IP: 192.168.22.56 Port: 49551 Auth-User: 1005 Auth-Realm: 192.168.11.30 MWI-Account: 1005(a)192.168.11.30 Call-ID: 001bd433-4c190002-d08bed92-50fab18f(a)192.168.22.57 User: 1010(a)192.168.11.30 Contact: "user" <sip:1010@192.168.22.57:5060 ;transport=udp;fs_nat=yes;fs_ path=sip%3A1010%40192.168.22.57%3A51471%3Btransport%3Dudp> Agent: Cisco-CP7941G/8.5.2 Status: Registered(UDP-NAT)(unknown) EXP(2010-05-02 12:46:04) Host: s-efrn2brk8zpcy IP: 192.168.22.57 Port: 51471 Auth-User: 1010 Auth-Realm: 192.168.11.30 MWI-Account: 1010(a)192.168.11.30 Call-ID: MjYzNmQ5YWYyMDkwMDc3M2UxYzY5Y2JkZGE4NTBhZWU. User: 1001(a)192.168.11.30 Contact: "3CXPhone" <sip:1001@192.168.22.58:3884 ;rinstance=74bf6bb1ef7db1 27;fs_nat=yes;fs_path=sip%3A1001%40192.168.22.58%3A3884%3Brinstance%3D74bf6bb1ef 7db127> Agent: 3CXVoipPhone 4.0.9878.0 Status: Registered(UDP-NAT)(unknown) EXP(2010-05-02 12:45:37) Host: s-efrn2brk8zpcy IP: 192.168.22.58 Port: 3884 Auth-User: 1001 Auth-Realm: 192.168.11.30 MWI-Account: 1001(a)192.168.11.30

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Build fails on Suse
by Matt White 03 May '10

03 May '10

I'm using the opensuse build service to create rpm's for Suse. The build fails with the following error below. I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions I: Expression compares a char* pointer with a string literal. Usually a strcmp() was intended by the programmer E: freeswitch stringcompare strings/apr_snprintf.c:1261 Any thoughts? I'm using the nightly snapshot. -M

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Accessing SQLITE core.db
by Saeed Ahmed 03 May '10

03 May '10

Hi, Is it safe to access sqlite db using PHP, when there are live calls on FS? I am just sending selects to 'channels' table to view live calls. Thanks

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mod_xml_cdr is not logging at all.
by Frank Church 02 May '10

02 May '10

Mod_xml_cdr is not logging anything at all. I have created an additional directory called logs, and it is still not working. cdr_csv is logging fine. Below is the current file, are there any faults in it? Which directory does {prefix} point to? My current verson is 17408m on Windows <configuration name="xml_cdr.conf" description="XML CDR CURL logger"> > <settings> > <param name="url" value="http://192.168.1.20:8101/"/> > <param name="log-dir" value="log/xml_cdr"/> > <param name="log-b-leg" value="false"/> > <param name="prefix-a-leg" value="true"/> > <param name="encode" value="true"/> > <param name="err-log-dir" value="log/xml_cdr"/> --> > </settings> > </configuration> > -- Frank Church ======================= http://devblog.brahmancreations.com

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Emacs/Vim - XML Editing Settings
by Aloysius Lloyd 02 May '10

02 May '10

Hi All, The default xml configuration files indent very nicely. What is the setting used to edit the default xml configuration files. Could you please share the settings either for emacs or vim. I prefer to use emacs. Thanks in advance. Lloyd

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Connect external phone
by Sean Holt 02 May '10

02 May '10

Hello list, I keep getting this authentication error when I attempt to connect a device from the outside. I followed the wiki that suggest creating a separate profile to handle external device connections. Here¹s my external xml file <profile name="gexternal5090">  <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${moh_uri}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="71.133.39.219"/> <param name="ext-sip-ip" value="71.133.39.219"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> I did a siptrace on the incoming connection attempt ------------------------------------------------------------------------ REGISTER sip:71.133.39.219:5090 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej Max-Forwards: 70 From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl To: <sip:1001@71.133.39.219> Call-ID: fo33ia7sk6qy9vnk6h CSeq: 75203 REGISTER Contact: <sip:1001@x.x.x.x:5090;line=20880> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 600 User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255) Content-Length: 0 ------------------------------------------------------------------------ send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108 From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl To: <sip:1001@71.133.39.219>;tag=Zc7mBaBN723Bp Call-ID: fo33ia7sk6qy9vnk6h CSeq: 75203 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="71.133.39.219", nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ Thoughts? Thanks Sean

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Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
by Nicola von Thadden 01 May '10

01 May '10

Hi, is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install). FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?) Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why. Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30. Conferences are also choppy. Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help. Anyone got an idea? Nico

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get a special var from accepted call
by Madovsky 01 May '10

01 May '10

Hi, I'd like to set a var (trunk ID) in the bridge that I can get from my SIP phone when the call is accepted. channel variables in set / export ? Thanks Franck

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