July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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.

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how to make web call on mod_curl using proxy ?
by Jack Nikolas 08 Jul '12

08 Jul '12

hi All, i am going to making web call using mod_curl in freeswitch, it is noticable that on my freeswitch for having internet access ,user have to set proxy, for example set proxy on the address like this: http_proxy=http://10.10.10.1:2343 now i want to running web call as the following way,but how can i set proxy on lua? web_url = "http://tycho.usno.navy.mil/cgi-bin/timer.pl" num_reads = 0 api = freeswitch.API() session:answer() while(session:ready() == true and num_reads < 10) do freeswitch.consoleLog("INFO","URL: " .. web_url .. "\n") raw_data = api:execute("curl", web_url) freeswitch.consoleLog("INFO","Raw data:\n" .. raw_data .. "\n\n") date_time = string.match(raw_data,"<BR>.-UTC",1) if (date_time == nil) then freeswitch.consoleLog("INFO","UTC date and time not found\n") else freeswitch.consoleLog("INFO","UTC date and time is '" .. date_time .. "'\n") time = string.gsub(date_time,".-(%d+:%d+:%d+).+","%1") freeswitch.consoleLog("INFO","Time is '" .. time .. "'\n\n") session:streamFile("phrase:simple_time:" .. time) end num_reads = num_reads + 1 session:execute("sleep","1000") end session:hangup() any help is welcome,

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Performance Testing
by Hanie Maghsoudy 08 Jul '12

08 Jul '12

Hi all, I searched for FreeSwitch call capacity, but most of the results wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's performance recently, or got a dramatic result in real environment? I tested call quality on this machine: Virtual FreeSwitch server OS: CentOS release 6.2 - x86_64 CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz Memory: 8 G After receiving incoming calls, FreeSwitch routed them to another sip server, without transcoding. The other server transmitted calls by playing an audio file. Meanwhile, I called an extension in FreeSwitch to test the call quality. The result was like this: 1000 Concurrent calls Call duration: 160s Call rate: 6 cps (just creating channels) Max used Memory: 1416M Max CPU load: 0.24 Max Network throughput (recv/send): 6711k/80k Quality: Good This test was taken before tearing down the channels. Then, I took another test to estimate calls per second, and it wasn't what I was expected! 150 Concurrent calls Call duration: 4s Call rate: 30 cps (creating and tearing down) Max used Memory: 1293M Max CPU load: 4.50 Max Network throughput (recv/send): 828k/60k Quality: Average And when I increase call rate to 50 cps: 1000 Concurrent calls Call duration: 4s Call rate: 50 cps (creating and tearing down) Max used Memory: 1730M Max CPU load: *29.9* Max Network throughput (recv/send): 1367k/202k Quality: Bad Why call per second is such a big problem? Did anyone get a better result on this? Thanks, Hanie

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Pushing CDR Information into Zeromq
by Neo Cheema 08 Jul '12

08 Jul '12

Hi all, I was hoping to find a way to push CDR information into a queue. Zeromq comes out as an obvious choice because mod_event_zmq already exits. However, I can't find a way to configure this module to push CDR info. Have any of you guys tried it? Regards

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Disable reinvite
by Richard Klingler 07 Jul '12

07 Jul '12

Hello Just switched this weekend from asterisk to freeswitch on freebsd 9.0. And so far I was able to implement basic dialplan for inbound/outbound calls (o; What I'm trying to find is a similar configuration option from asterisk sip.conf where individual user agents can have SIP reinvite disbaled with "canreinvite=no". Fomr the Freeswitch documentation it states that all RTP streams are passed through by default, but I still see SIP clients trying to send UDP streams to my box directly. Also setting "<param name="inbound-proxy-media" value="true"/>" in sofia.conf.xml doesn't solve this problem. I just want the UDP streams frmo outside are only allowed from my SIP trunk provider. thanx in advance richard

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sensibility of voice level for Mod_pocketsphinx
by Yuriy Nasida 07 Jul '12

07 Jul '12

Hi guys, Whether somebody know how to lower sensibility of voice level for Mod_pocketsphinx. I play with "threshold" in pocketsphinx.conf.xml but I have not any effect.the Mod_pocketsphinx begins to generate event == "begin-speaking" when I don't say anything... Also i would like do NOT break the current playing of session:streamFile when events of Mod_pocketsphinx is being generated. Is it possible ? Also can somebody explain that this parameters means ? I understand approximately but I would like to know for certain. I have found nothing in wiki. <param name="threshold" value="400"/> <param name="silence-hits" value="25"/> <param name="listen-hits" value="1"/> <param name="auto-reload" value="true"/>    Thanks.

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Problem with Polycom behind firewall with freeswitch
by Akhil Chandra 07 Jul '12

07 Jul '12

Hi all, I come from a asterisk background and having trouble keeping the firewall hole opened up for Polycom registering to freeswitch server. I am running freeswitch ( with bluebox UI) and trying to register a Polycom behind a router. I have exhausted all the options in sip_profiles directory for the interface I am using. The issue I am seeing is that the freeswitch sends the OPTIONS message to the internal IP ie ( 192.168.X.X) rather than the public IP, behind which the Polycom phones are. These phones register fine with asterisk in the stated scenario. I have tried changing these options in the xml file but still the OPTIONS message ( as seen in tcpdump) is sent to 192.168.X.X <param name="nat-options-ping" value="true"/> <param name="all-reg-options-ping" value="true"/> <param name="apply-nat-acl" value="rfc1918.auto"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> Any help appreciated !! .. Thanks in advance. Regards, Akhil

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how to prevent of changing the value of nonce ttl sip auth ?
by Stefan Shortner 07 Jul '12

07 Jul '12

HI, i find out every time we want to register on sipclient the sip_nonce_auth get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have <param name="nonce-ttl" value="60"/> it means every 60sec the value of sip_nonce_auth for each sipclient was chnaged , and if the value of nonce-ttl is less than the value of parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced

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how to prevent of changing the value of nonce ttl sip auth ?
by Jack Nikolas 07 Jul '12

07 Jul '12

----- Forwarded Message ----- From: Jack Nikolas <jack.nikolas(a)ymail.com> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, 6 July 2012, 11:20:52 Subject: Fw: how to prevent the changes of nonce ttl sip auth ? ----- Forwarded Message ----- From: Jack Nikolas <jack.nikolas(a)ymail.com> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, 6 July 2012, 10:12:43 Subject: how to prevent the changes of nonce ttl sip auth ? ----- Forwarded Message ----- From: Jack Nikolas <jack.nikolas(a)ymail.com> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, 6 July 2012, 10:10:50 Subject: how to prevent the changes of nonce ttl sip auth ? HI, i find out every time we want to register on sipclient the sip_nonce_auth get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have <param name="nonce-ttl" value="60"/> it means every 60sec the value of sip_nonce_auth for each sipclient was chnaged , and if the value of nonce-ttl is less than the value of parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced

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Load Balancing
by Hanie Maghsoudy 06 Jul '12

06 Jul '12

Dear all, I tried OpenSIPs for load balancing. I works just fine, but as it is mentioned here,<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#CDRs>OpenSIPs sends requests from itself. How about Ultramonkey? Does it change the source too, or it just manipulate destination of registration packets? Thanks, Hanie <http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#CDRs>

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