June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FaxxBochs hackers?
by Sean Devoy 11 Jul '12

11 Jul '12

Has anyone investigated the new FaxxBochs offering? It appears to handle the analog fax machine locally, then send the fax as a data TCP connection to the server (as a TIFF or JPG I would guess). It totally removes the VOIP problem with fax machines and is a clever solution. $30 a month unlimited in and out is reasonable, but I would rather have that income myself. It would be great if I could resell it and make a cut, but they require 75 device sales the first year. If someone knows anything about "reconfiguring" one of these, I would love to read about it. Sean

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Attended transfer via ESL
by Limit 10 Jul '12

10 Jul '12

I'm trying to make an attended transfer through EventSocket, using sendmsg with att_xfer command, but having problems with it. ESL function: execute("att_xfer", args, b_uuid); It creates new calls as normal dialplan att_xfer, but problem occured when one of three sides hangs up. Sometimes it goes as planned, but more often all channels hangup automatically. att_xfer called from dialplan work as needed, looks like there is some problem with EventSocket call specifically . Maybe I must set some variables before to use it properly? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Different Voices for FS
by toddb＠toddbailey.net 09 Jul '12

09 Jul '12

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spandsp error reading frame
by Jean-Marc Hyppolite 09 Jul '12

09 Jul '12

Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/.@...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/.@...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/.@... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/.@...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/.@...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc. N.B. Sorry if some people received this e-mail twice. I am not sure this email went through the first time.

3 3

FreeSWITCH TLS with StartSSL Certificate
by sunsus 08 Jul '12

08 Jul '12

Hello To day I tried to add a Free StartSSL Class 1 Certificate to a FreeSWITCH installation. Here I will share the script on how to generate the Certificate Request: #!/bin/sh CONFDIR=/usr/local/freeswitch/conf/ssl DAYS=2190 KEY_SIZE=2048 TMPFILE="/tmp/fs-ca-$$-$(date +%Y%m%d%H%M%S)" COMMON_NAME="FrwwSWICH VOIP" ALT_NAME="DNS:sip.freeswitch.org" ORG_NAME="FreeSWICHT" OUTFILE="agent.pem" umask 037 generate_request() { local val="" echo "Generating new request..." echo echo "--------------------------------------------------------" echo "CN: \"${COMMON_NAME}\"" echo "ORG_NAME: \"${ORG_NAME}\"" echo "ALT_NAME: \"${ALT_NAME}\"" echo echo "Certificate filename \"${OUTFILE}\"" echo echo "[Is this OK? (y/N)]" read val if [ "${val}" != "y" ] && [ "${val}" != "Y" ]; then echo "Aborted" return 2 fi sed \ -e "s|%CN%|$COMMON_NAME|" \ -e "s|%ALTNAME%|$ALT_NAME|" \ -e "s|%ORG%|$ORG_NAME|" \ "${CONFDIR}/CA/config.tpl" \ > "${TMPFILE}.cfg" || exit 1 echo ${KEY_SIZE} openssl req -new -out "${TMPFILE}.req" \ -newkey rsa:${KEY_SIZE} -keyout "${TMPFILE}.key" \ -config "${TMPFILE}.cfg" -nodes -sha1 >/dev/null || exit 1 echo cat ${TMPFILE}.req echo echo "go to http://www.startssl.com/ and generate a certificate" echo "past certificate:" while read LINE do echo $LINE >> ${TMPFILE}.crt if [ "$LINE" = "^A" ];then break fi done echo "other processing continues " # openssl x509 -req -CAkey "${CONFDIR}/CA/cakey.pem" -CA "${CONFDIR}/CA/cacert.pem" -CAcreateserial \ # -in "${TMPFILE}.req" -out "${TMPFILE}.crt" -extfile "${TMPFILE}.cfg" \ # -extensions "${EXTENSIONS}" -days ${DAYS} -sha1 >/dev/null || exit 1 cat "${TMPFILE}.crt" "${TMPFILE}.key" > "${CONFDIR}/${OUTFILE}" wget http://www.startssl.com/certs/sub.class1.server.ca.pem wget http://www.startssl.com/certs/ca.pem cat sub.class1.server.ca.pem ca.pem >> ${CONFDIR}/cafile.pem rm -f sub.class1.server.ca.pem ca.pem rm "${TMPFILE}.cfg" "${TMPFILE}.crt" "${TMPFILE}.key" "${TMPFILE}.req" echo "DONE" } remove_startssl() { echo "Removing StartSSL" if [ -d "${CONFDIR}/agent.pem" ]; then rm "${CONFDIR}/agent.pem" fi echo "DONE" } OUTFILESET="0" command="$1" shift while [ $# -gt 0 ]; do case $1 in -cn) shift COMMON_NAME="$1" ;; -alt) shift ALT_NAME="$1" ;; -org) shift ORG_NAME="$1" ;; -out) shift OUTFILE="$1" OUTFILESET="1" ;; -days) shift DAYS="$1" ;; esac shift done case ${command} in create_request) EXTENSIONS="request" generate_request ;; remove) echo "Are you sure you want to delete the StartSSL Certificate? [YES to delete]" read val if [ "${val}" = "YES" ]; then remove_startssl else echo "Not deleting CA" fi ;; *) cat <<-EOF $0 <create_request|remove> [options] * commands: remove - Remove StartSSL create_request - Create a new certificate request for startSSL * options: -cn Set common name -alt Set alternative name (use prefix 'DNS:' or 'URI:') -org Set organization name -out Filename for new certificate (create only) -days Certificate expires in X days (default: 365) EOF exit 1 ;; esac Everything seams to work, expect the validation of a SNOM phone. Does any one know how to tell FreeSWITCH to publish the correct ca bundel and certificate track. Because the CA Certificate of Start SSL is included in the SNOM: regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-TLS-with-StartSSL-… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

how to get the content of sip header ?
by Samira Mh 03 Jul '12

03 Jul '12

hi, how to get the content of sip header in lua?

4 6

Simple SPA8800 FXO connection to FreeSwitch
by Robin Gilks 01 Jul '12

01 Jul '12

Greetings I'm a noob with FS (and VoIP generally!!) and am really struggling with getting 2 way comms between a PABX and a soft-phone using FS and an SPA8800. Calls from the PABX are OK, the FXO ports have registered with FS and when I dial the FXO extension I get the dial tone, dial the number for the soft-phone and it all works fine. The problem is connections *TO* the PABX. The SPA8800 AFAIK has a separate SIP port for each of its FXO ports. My understanding of how it should work is as follows: I have to have 4 gateways defined in a file in the sip-profiles/externals directory, each with a different "sip-port" line. I'm testing with 2 to start with: # cat cicsco_fxo.xml <include> <gateway name="fxoline3"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="realm" value="172.27.1.20"/> <param name="sip-port" value="5260"/> <param name="register" value="false"/> </gateway> <gateway name="fxoline4"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="realm" value="172.27.1.20"/> <param name="sip-port" value="5360"/> <param name="register" value="false"/> </gateway> I have to use mod_distributor to select one of the gateways that define the SIP ports. Thats is configured in the autoload_configs directory. # cat distributor.conf.xml <configuration name="distributor.conf" description="Distributor Configuration"> <lists> <list name="cisco_fxo" total-weight="2"> <node name="fxoline3" weight="1"/> <node name="fxoline4" weight="1"/> </list> </lists> </configuration> Finally, I use a dialplan to access this distribution of gateways when a 4 digit number starting with 8 is dialed from the soft-phone. # cat 33_ciscofox.xml <include> <extension name="pabx"> <condition field="destination_number" expression="^(8\d{3})$"> <action application="bridge" data="sofia/gateway/${distributor(cisco_fxo)}/$1"/> </condition> </extension> </include> Result expected - it all works. Result I get is: freeswitch(a)PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002(a)172.16.164.33 [d0fc47e6-c1a0-11e1-881d-f70f560259d2] 2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing 1002 <1002>->8569 in context default 2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway '' 2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A [CS_NEW] 2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] 2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed. Cause: INVALID_GATEWAY 2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup sofia/internal/1002(a)172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY] 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session 11 (sofia/internal/1002(a)172.16.164.33) Ended 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/1002(a)172.16.164.33 [CS_DESTROY] So what is invalid about the gateway? Cheers -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. It is subject to copyright, is confidential and may be the subject of legal or other privilege, none of which is waived or lost by reason of this transmission. If you are not an intended recipient, you may not use, disseminate, distribute or reproduce such email, any attachments, or any part thereof. If you have received a message in error, please notify the sender immediately and erase all copies of the message and any attachments. Unfortunately, we cannot warrant that the email has not been altered or corrupted during transmission nor can we guarantee that any email or any attachments are free from computer viruses or other conditions which may damage or interfere with recipient data, hardware or software. The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------

3 3

Strange issue setting variables for mod_xml_cdr
by Muhammad Naseer Bhatti 01 Jul '12

01 Jul '12

I have gone nuts trying to figure out what is going on. Using xml_cdr to post CDR(s) to webserver, The Bridge is done outside the lua script, and all vars are set in lua and exported. .... session:setVariable("import", "gateway") session:setVariable("import", "admin_rate_group") session:setVariable("import", "sell_rate") session:setVariable("import", "cost_rate") .... The problem is that if I set sell_rate and cost_rate both variables together, who ever is in the end disapears from the XML POST. If I change their order, for example set cost_rate first and then sell_rate, then sell_rate works fine and cost_rate goes off. Looks like some sort of variable overlapping. I am HEAD with latest/latest. freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n cost_rate ".. cost_rate .." \n"); freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n sell_rate ".. sell_rate .." \n"); This works fine. There is data in these two variables. I am setting up these session variables along with other variables for the bridge. log-b-leg is false in the config for xml_cdr. http://pastebin.freeswitch.org/19408 is the complete PB for the call log. http://pastebin.freeswitch.org/19409 is the complete XML CDR debug. cdr_csv shows the same "Bhatti",*"2.10000","0.19280"*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be292014-c0b8-11e1-ae74-2bc60749f960","","","","" "Bhatti",*"2.10000",""*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be2782fe-c0b8-11e1-ae6f-2bc60749f960","be292014-c0b8-11e1-ae74-2bc60749f960","","","" Sometimes it works sometimes it wont. All the variables are seen in the call log. Am I doing something wrong or we hit some sort of a bug? Thank.

1 1

Relatively new FreeSwitch user, calls from outside not terminated
by Thomas Brettinger 01 Jul '12

01 Jul '12

Hi, I started toying around with FreeSwitch on PFsense a while ago, and now did a fresh install on a machine inside my lan. Setup is: FS --- PFsense (Firewall, NAT) --- Internet --- sipgate.de Routing inside and outside works like a charm. However, I have the issue that if a call is hung up on a local phone, FreeSwitch doesn't hang up on the sipgate.de leg. It doesn't make a difference if the call was originated from inside or outside. The call is left open for an unknown time and freeswitch doesn't seem to notice that the inside has hung up (at least, nothing appears in the sip trace or log). As soon as the remote party hangs up, siptrace and logfile look as they probably should. Does anyone have an idea where I should start looking for this issue? It *could* have something to do with the firewall between, however I don't believe so, as the call is hung up on the local network and freeswitch doesn't seem to notice. Oh, and I tried this with Cisco 69xx (SIP) and Siemens GIgaset 501V (SIP) phones. Any hints appreciated. Thanks in advance, Thomas

2 1

separate channel playback
by Mi Ke 01 Jul '12

01 Jul '12

Dear FreeSwitchers, Is it possible to select a particular channel (left or right) when I play stereo audio file to the call leg or conference? Thanks / MiKe

2 1