September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

A Chance To "Give Back" To FreeSWITCH
by Michael Collins 27 Sep '11

27 Sep '11

Hello all, I have a few FreeSWITCH-related projects with which I could use a hand. We have some non-glamorous janitorial items that need some attention. Specifically I could use assistance with a few freeswitch.org and wiki.freeswitch.org items. Some research/digging for answers is involved. If you have PHP skills those are a plus. Please email me off list if you are able to devote a few hours to some projects that I have pending. We will also discuss some of these on the FreeSWITCH conference call tomorrow. Thanks for giving back <http://www.freeswitch.org/node/339>. -Michael Collins

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Any DID
by Billy L. Irwin 27 Sep '11

27 Sep '11

Hi All, I am running Freeswitch with Blue.Box as a front end. Is there a way to add an inbound route that would support any DID as I did within Asterisk? I don't want to have to add all our DIDs to users right off the bat? Thanks, Billy

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Gateway Authentication
by Chad Vogel 27 Sep '11

27 Sep '11

Hello, I'm trying to make the move from Asterix, but I'm running into some difficulties. I'm try to bridge a call using our gateway however it doesn't work. In wireshark I can see I'm getting an SIP 401 Unauthorized error with a WWW-Authenticate header, after FS send the INVITE message to the gateway. However FS doesnt seem to respond to the request for Authentication. Asterix responds correctly however I cant seem to make FS to do the same. Any help would be appreciated INVITE sip:+15618911806@4.55.35.60:5070 SIP/2.0 Via: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKB49SZQHrgaaKc Max-Forwards: 8 From: "LyonL" <sip:+14142211800@207.67.30.226:5060>;tag=eZe8gcQgXXv5c To: <sip:+15618911806@4.55.35.60:5070> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f CSeq: 17931324 INVITE Contact: <sip:1-F2la9@207.67.30.226:5060;transport=udp;gw=level3> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 227 X-FS-Support: update_display Remote-Party-ID: "LyonL" <sip:+14142211800@207.67.30.226:5060>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6 s=FreeSWITCH c=IN IP4 10.126.200.6 t=0 0 m=audio 17944 RTP/AVP 0 8 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 207.67.30.226;received=207.67.30.226;branch=z9hG4bKB49SZQHrgaaKc;rport=42534 From: "LyonL" <sip:+14142211800@207.67.30.226:5060>;tag=eZe8gcQgXXv5c To: <sip:+15618911806@4.55.35.60:5070>;tag=SD6soqf99-1367649635-1316534779161 Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f CSeq: 17931324 INVITE WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks" Content-Length: 0 <include> <extension name="4142211800"> <condition field="destination_number" expression="^(\+?1)?(4142211800)$"> <action application="set" data="effective_caller_id_name=LyonL"/> <action application="set" data="effective_caller_id_number=+14142211800"/> <action application="bridge" data="sofia/gateway/level3/+15618911806"/> </condition> </extension> </include> <include> <gateway name="level3"> <param name="apply-inbound-acl" value="level3"/> <param name="username" value="1-F2la9"/> <param name="password" value="password"/>> <param name="realm" value="BroadWorks"/> <param name="proxy" value="4.55.35.60:5070"/> <param name="from-domain" value="207.67.30.226:5060"/> <param name="dtmf-type" value="rfc2833"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="register" value="false"/> </gateway> </include> <profile name="external">   <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>    <param name="local-network-acl" value="localnet.auto"/> <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="user-agent-string" value="FreeSWITCH"/>  <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>   <param name="tls" value="$${external_ssl_enable}"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="$${external_tls_port}"/>  <param name="tls-cert-dir" value="$${external_ssl_dir}"/>  <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile>

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large block of DID's
by Chad Vogel 27 Sep '11

27 Sep '11

I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers) Chad

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Error loading mod_shout
by William Moore 27 Sep '11

27 Sep '11

I get the following error when I load mod_shout through the cli. freeswitch@internal> load mod_shout +OK Reloading XML -ERR [module load file routine returned an error] 2011-09-23 00:31:10.654023 [INFO] switch_time.c:1028 Timezone reloaded 530 definitions freeswitch@internal> 2011-09-23 00:31:10.654023 [CRIT] switch_loadable_module.c:1271 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: curl_easy_setopt** Any ideas? -- William J. Moore Statira Systems 611 Caroline St Fredericksburg, VA http://www.statirasystems.com

2 1

TLS issue
by Ben Naylor 27 Sep '11

27 Sep '11

Hello Freeswitch community! I was wondering if someone could help me with an issue I am experiencing when using TLS to secure SIP traffic? Our current test environment consists of a Freeswitch test server, a wireless access point, and 2 x iPhones using SIP client, all connected to the same closed off VLAN. The Freeswitch server is set to bypass media=true, so that the RTP traffic flows directly between the iPhones. When testing a call using standard SIP/RTP, the call is successful, and terminates correctly when one of the devices hangs up the call. The problem occurs when using TLS, and trying to terminate the call. The call establishes successfully, but when either of the handsets hangs up, the other remains connected to the call. This happens when hanging up the call from both the calling and the called handset. One thing I noticed when looking at the console output, is that two messages appear along the lines of 'unknown Sip packet' (apologies for the vagueness, I am not currently in front of the test server). These appear when the call is terminated on one of the hand sets. Has anyone had a similar experience when using TLS in this type of scenario? Thanks for the help! Regards Ben

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Mitel 52xx Broadsoft SLA/SCA Issues
by aromberg＠gmail.com 27 Sep '11

27 Sep '11

Here's the quick synopsis: Freeswitch on CentOS 6 freshly installed 2 days ago. Freeswitch was grabbed via git. Phones are Mitel 5220 SIP phones that are in SIP mode. Mitel has a Broadsoft implementation called SIP_SCA in the device registration. There is 1 Cisco 7960 but its not configured at all for the shared extension. I can't get it to work. I *think* I have it narrowed down to the presence ID being different on a per-phone basis. The LEDS show the correct call status (on-hook, ringing, on hold, etc). However, when you press the shared line from another phone, it dials the extension again and goes straight to voicemail/busy instead of seizing the line. All extensions are configured correctly (calling the shared extension from the Cisco results in both phones ringing and able to pick up the call initially) Logfiles are located at: http://iis.powertogrow.com/fs Each one has the same actions: Mitel phone 1 (192.168.1.106/extention 1002) calls Cisco Phone 2 (192.168.1.103/#1000) on Shared Line 1003. Cisco answers. Mitel Phone 1 puts call on hold. Mitel phone 2 (192.168.1.100/#1001) tries to pick up Shared Line 1003 and then hangs up when it hears the voicemail system. It normally does this twice. Then Mitel phone 1 picks up Shared Line 1003 and ends it. No matter what order the steps are done with whatever phone (well, except the Cisco doesn't have a shared line), the results are the same. Picking up the shared line results in a forward to the VM system Thanks, Adam

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Call Hangup with INCOMPATIBLE_DESTINATION cause
by Stephen Wilde 26 Sep '11

26 Sep '11

Hi all, I have some calls dropped with INCOMPATIBLE_DESTINATION cause. I have checked the SDP of the bridged legA and legB and to me appear ok (both in G729). What I see in the log is: http://pastebin.freeswitch.org/17396 Attached to this email an example of captured trace. Any advice on that? Stephen

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Call forward
by baskar 26 Sep '11

26 Sep '11

Hi All, I try to make call forward form once extension to another extension. Example: I have three extensions 4100, 4107, 4102 Call forwarded from 4107 to 4102 but it directly go to voicemail. If i cancel call forward it directly reaches the extension. Can any one help to resolve this issue. I have paste my log in http://pastebin.freeswitch.org/17386. Thanks in advance. N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-forward-tp6822324p682232… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Auto adjust Change port RTP
by MarhBere 26 Sep '11

26 Sep '11

Hi, I have an issue with extension (eyebeam) behind nat. Scenario: The call Incoming From ITSP and is bridged with the a user registered; The RTP arrive to FS from register user, but don't have the auto adjust IP/Port. So only had silence in both points. IF the phone Hold the call and Unhold the call. FS now yet change the RTP to correct private IP/Port and call is now with bidirectional audio. I red that is hapened in old release : http://freeswitch-users.2379917.n2.nabble.com/No-RTP-destination-auto-corre… But I running the last Head trunk. "FreeSWITCH Version 1.0.head (git-8df1872 2011-09-19 13-51-41 -0400)" SO: CentOs 5. 64x Someone know what is wrong? PD: Sorry for my traduction. Thanks, Marcelo Bereterbide. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Auto-adjust-Change-port-RTP-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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