- FreeSWITCH-users - FreeSWITCH Mailing Lists

mobile phone clients / fs cluster
by Phillip Jones 03 May '10

03 May '10

Hi there, I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy. Outbound (terminating) traffic goes directly from each FS box. All my home office phones get calls no matter which box they are registered with or which box the call comes in on - thanks Anthony. However my SIP client on various iPhone/Androids etc only receive calls that originate on the box on which they are registered. Looking at the SIP trace - when the call comes in on the 'wrong' box, the invites to these SIP clients do not even get a response. Presumably because the 3G network has no idea who this new IP in the "from address" is, who trying to contact them. Question is, is there a way around this - our will I have start routing all terminating traffic out through the proxy also. Any insight appreciated. Thanks! Pj

3 4

How to send "waiting call" tone to remote side
by Yehavi Bourvine 03 May '10

03 May '10

Hello, At the default dialplan of freeswitch, when there is a waiting call the remote side hears a normal "ring" tone. I would like to change it to send a "call waiting" ring tone. What is the official way of doing so? I can think of two ways: - sending 182 instead of 180. - send 180 with early media and play the ring tone. I've tried the second method. It works inside our PBX and at the adjacent telco PBX. On other telco PBXes it doesn't work (silence). Before I call the telco I would like to know what is the common way of doing so. Thanks! __yehavi:

3 5

absolute_codec_string vs codec_string
by Ken Fulmer 03 May '10

03 May '10

Would someone please clarify the difference between the following parameters? I've seen the Wiki page but still don't understand the difference. 1. absolute_codec_string 2. codec_string Thanks, Ken Fulmer

2 2

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

1 0

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

1 0

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

3 6

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

2 1

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

3 3

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

4 8

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

2 2