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IVR + Cell Phone Weirdness
by George Sapak 14 Feb '12

14 Feb '12

I am having a weird issue with cell phones reaching the IVR, it seems the IVR only picks up the first number dialed not all three, this only happens from a cell. land lines are fine this is what I have to direct calls to internal extensions: <entry action="menu-exec-app" digits="/(^\*\d{3}$|^\d{3}$)/" param="transfer $1 XML default"/> Any help is greatly appreciated, Thank You, George

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FreeSWITCH Crash - How to Troubleshoot
by Lloyd Aloysius 14 Feb '12

14 Feb '12

Hi All, I was running the following version for a long time and never have any issue *freeswitch@internal> version FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)* * * *---------------------* * * Two days ago Brian suggest to update to a latest git for call pickup to working. So I did update. My New version is as follow * * *FreeSWITCH Version 1.0.head (git-eeeaf41 2012-02-10 17-27-50 -0600)* Today morning FreeSWITCH crash. How to troubleshoot this problem. How to find the reason .. why this crash? Thanks and regards, LLoyd

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SDP with 2 m=audio lines
by Sergey Okhapkin 14 Feb '12

14 Feb '12

I noticed that recent freeswitch versions could send SDP with 2 m=audio lines. Per RFC 3388, the behavior is undefined this case. I don't know how to reproduce the problem, I see such packets coming from my customers. Here is an example: INVITE sip:xxxxxxxxxxxx@sip.callwithus.com SIP/2.0 ... User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-eeeaf41 2012-02-10 17-27-50 -0600 .... v=0 o=FreeSWITCH 1329167961 1329167962 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 24598 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 24598 RTP/AVP 98 0 8 3 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Hmm, I just realized that this SDP is buggy - 2 audio streams on the same port, with common codecs, but with different ptime :-)

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Inline execution concept
by virendra bhati 14 Feb '12

14 Feb '12

Hi List, I am the new guy for FreeSwitch. I am not clear what is the used of Inline execution. As per doing some tweak I found that Inline is used for set any variable for channel base like in asterisk we used double underscore( __ ). Am i right on this concept or not? Another question is how to include context in dialplan in XML ? I am reading FreeSwitch 1.0.6 for more details but still these question is not clear for me. Please help me to find out the answer of these questions. -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2

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segmentation fault on startup
by Darcy 14 Feb '12

14 Feb '12

I have three freeswitches running on public IPs, not behind a router. There was a power hit on site and all servers were restarted but none of the freeswitches started, when I tried to manually start them, I got a segmentation fault. I had to rename core.db on each system to get them to restart. Is this a known issue. One of these switches had a git update 1 week ago, the other two are various vintages but run flawless, until last night of course. The longest running one is about 14 months. Any input or thoughts would be appreciated. Darcy

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bypass media only if on same subnet?
by Tim St. Pierre 14 Feb '12

14 Feb '12

Hello, In order to eliminate NAT issues, we have built routable tunnels to our larger customers so that their phones are on a private subnet that is routable to a private subnet at our datacenter. Each Freeswitch system has a profile called nonat that uses an interface and address bound to this network. The remaining customers register to the "internal" profile, which is bound to a public IP address. At the moment, everything works just great - no nat issues, instant failover between the primary and secondary (shared registrations in DB), but all media flows through our network. If someone in office A calls another phone in office A, I would like FS to instruct the phones to send their media direct. The addresses and ports are all correct in this case. If someone in office A calls a phone in office B, I want Freeswitch to stay in the media path, as these two offices are not routable to each other, even though each is routable to Freeswitch. Is there a way to set up a profile (or dialplan) so that FS will bypass media only if the two endpoints are on the same subnet? An ACL isn't really the right thing, since it would require an exponential number of ACLs. Also, many calls go to ring groups, where several phones ring and we don't know which one will answer until it actually does (eliminating some sort of dialplan code using the rtp variables). Is this possible?

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questions about modules.conf
by Josh 14 Feb '12

14 Feb '12

According to the Installation/Build Guide (http://wiki.freeswitch.org/wiki/Installation_Guide) modules.conf gets generated during the execution of bootstrap.sh. Should I assume that the options specified in this file - both in terms of what modules are included as well as whether or not they are enabled - depends on my own configuration (the presence of required packages & libraries) as well as the presence of the source code/makefile for a particular module or is the content of this file "predefined" and filled in with a set of "standard" modules regardless of what bootstrap has found (I hope the answer is the former, but need to make absolutely sure)? Also, the same wiki page recommends that I also fetch freeswitch-contrib.git as well as freeswitch-sample-configs.git, but it is not clear to me whether I should dump the contents of these two projects within the main FS tree, where freeswitch.git was placed, or whether I should compile/build these separately - in a completely separate tree? Thanks!

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Polycom BLF w/ OpenSIPS and FreeSWITCH not working
by Vik Killa 14 Feb '12

14 Feb '12

I am unable to get BLF working when I have OpenSIPS in front of FreeSWITCH Would anybody know what is going wrong? 72.43.207.158 is OpenSIPS (test1.voip.demo.i-evolve.net) 72.43.207.155 is FS Here is a SIP trace take from FS on a REGISTER/NOTIFY/SUBSCRIBE: recv 725 bytes from udp/[72.43.207.158]:5060 at 13:30:15.670418: ------------------------------------------------------------------------ REGISTER sip:72.43.207.155:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKf54f.188e1555.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKf59d03a4266B7F41 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net> CSeq: 1 REGISTER Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 30 Expires: 3600 Content-Length: 0 Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 707 bytes to udp/[72.43.207.158]:5060 at 13:30:15.672807: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKf54f.188e1555.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKf59d03a4266B7F41 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net>;tag=B3eFSHXS8Q0vN Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 freeswitch@internal> freeswitch@internal> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="test1.voip.demo.i-evolve.net", nonce="079dfd86-5710-11e1-af11-2f1196b916eb", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 995 bytes from udp/[72.43.207.158]:5060 at 13:30:15.692185: ------------------------------------------------------------------------ REGISTER sip:72.43.207.155:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKc54f.4c7b9b9.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bK7a3b5bcD257C8B9 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net> CSeq: 2 REGISTER Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Authorization: Digest username="1000", realm="test1.voip.demo.i-evolve.net", nonce="079dfd86-5710-11e1-af11-2f1196b916eb", qop=auth, cnonce="CNyXC4dNuYJCV0G", nc=00000001, uri="sip:test1.voip.demo.i-evolve.net", response="4ff00198d12440fbc2aa14dcf9fb213b", algorithm=MD5 Max-Forwards: 30 Expires: 3600 Content-Length: 0 Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 711 bytes to udp/[72.43.207.158]:5060 at 13:30:15.713022: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bKc54f.4c7b9b9.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bK7a3b5bcD257C8B9 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=37656326-ABE8AE0B To: <sip:1000@test1.voip.demo.i-evolve.net>;tag=cc87tceX50pFH Call-ID: 2673bb72-3f195707-51caae10(a)192.168.99.21 CSeq: 2 REGISTER Contact: <sip:1000@72.43.207.130:1666>;expires=3600 Date: Tue, 14 Feb 2012 13:30:15 GMT User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Path: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> Content-Length: 0 ------------------------------------------------------------------------ send 964 bytes to udp/[72.43.207.158]:5060 at 13:30:15.827086: ------------------------------------------------------------------------ NOTIFY sip:1000@72.43.207.130:1666 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.155;rport;branch=z9hG4bK6Saj311cpBygK Route: <sip:72.43.207.158;lr;received=sip:72.43.207.130:1666> Max-Forwards: 70 From: <sip:1000@test1.voip.demo.i-evolve.net>;tag=DN10v7y029c2c To: <sip:1000@test1.voip.demo.i-evolve.net> Call-ID: df13f894-d1b2-122f-6aa8-928cc183345f CSeq: 24277043 NOTIFY freeswitch@internal> freeswitch@internal> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer freeswitch@internal> freeswitch@internal> Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:1000@test1.voip.demo.i-evolve.net ------------------------------------------------------------------------ recv 543 bytes from udp/[72.43.207.158]:5060 at 13:30:15.847271: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 72.43.207.155;received=72.43.207.155;rport=5060;branch=z9hG4bK6Saj311cpBygK From: <sip:1000@test1.voip.demo.i-evolve.net>;tag=DN10v7y029c2c To: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=BEC991BF-465E06A8 CSeq: 24277043 NOTIFY Call-ID: df13f894-d1b2-122f-6aa8-928cc183345f Contact: <sip:1000@72.43.207.130:1666> Record-Route: <sip:72.43.207.158;lr;ftag=DN10v7y029c2c> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ freeswitch@internal> recv 796 bytes from udp/[72.43.207.158]:5060 at 13:30:16.653565: ------------------------------------------------------------------------ SUBSCRIBE sip:1004@72.43.207.155:5060 SIP/2.0 Record-Route: <sip:72.43.207.158;lr;ftag=D90A32FE-DB2A2443> Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bK9c9.5db2f931.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKe4c1b0d5A735EFD6 From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 To: <sip:1004@test1.voip.demo.i-evolve.net> CSeq: 1 SUBSCRIBE Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 Contact: <sip:1000@72.43.207.130:1666> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 30 Expires: 3600 Content-Length: 0 X-AUTH-IP: 72.43.207.130 ------------------------------------------------------------------------ send 864 bytes to udp/[72.43.207.158]:5060 at 13:30:16.656893: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 72.43.207.158;branch=z9hG4bK9c9.5db2f931.0 Via: SIP/2.0/UDP 72.43.207.130:1666;branch=z9hG4bKe4c1b0d5A735EFD6 Record-Route: <sip:72.43.207.158;lr;ftag=D90A32FE-DB2A2443> From: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 To: <sip:1004@test1.voip.demo.i-evolve.net>;tag=P0syeQe6vUN3 Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 CSeq: 1 SUBSCRIBE Contact: <sip:1004@72.43.207.155:5060> Expires: 3600 User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ send 1300 bytes to udp/[72.43.207.130]:1666 at 13:30:16.731407: ------------------------------------------------------------------------ NOTIFY sip:1000@72.43.207.130:1666 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.155;rport;branch=z9hG4bK723a5vjgKmm3e Max-Forwards: 70 From: <sip:1004@test1.voip.demo.i-evolve.net>;tag=P0syeQe6vUN3 To: "Jackie C." <sip:1000@test1.voip.demo.i-evolve.net>;tag=D90A32FE-DB2A2443 Call-ID: 90f25b7b-2a54ccd4-69272b31(a)192.168.99.21 CSeq: 165317323 NOTIFY Contact: <sip:1004@72.43.207.155:5060;transport=udp> User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: presence Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/xpidf+xml Content-Length: 460 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <status> <note>Registered</note> </status> <presentity uri="sip:1004@test1.voip.demo.i-evolve.net;method=SUBSCRIBE" /> <atom id="sip:1004@test1.voip.demo.i-evolve.net"> <address uri="sip:mod_sofia@72.43.207.155:5060;user=ip" priority="0.800000"> <status status="" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> -----------------------------------------------------------------------

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mod_callcenter overloading database when using MySQL via unixODBC
by Vik Killa 14 Feb '12

14 Feb '12

I have 2 FS servers sharing a mysql database via ODBC. There are 2 queues loaded in mod_callcenter and there is absolutely zero activity around mod_callcenter. The mysql.log is showing many queries a few times per second like: "select 1" "SELECT queue,uuid,session_uuid,cid_number,cid_name,joined_epoch,(1329162473-joined_epoch)+base_score+skill_score AS score, state, abandoned_epoch FROM members WHERE state = 'Waiting' OR state = 'Abandoned' OR (serving_agent = 'ring-all' AND state = 'Trying') ORDER BY score DESC" Even if i run "callcenter_config queue unload support1@default" to unload the queues from both FS servers, the mysql.log still doing the queries. The only way the queries stop is if i do "unload mod_callcenter" Here is my callcenter.conf: <configuration name="callcenter.conf" description="Call Center Module"> <settings> <param name="odbc-dsn" value="freeswitch:freeswitch:freeswitch"/> </settings> <queues> <queue name="support1@default"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="local_stream://moh/8000"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="max-wait-time-with-no-agent-time-reached" value="5"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> <queue name="support2@default"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="local_stream://moh/8000"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="max-wait-time-with-no-agent-time-reached" value="5"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> </queues> </configuration>

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Re: [Freeswitch-users] Rotate cdr-records as root user (when FS is running from freeswitch user).
by Henrik Aagaard Sørensen 14 Feb '12

14 Feb '12

I just tested the killall -HUB freeswitch as well, and the cdr records does not get rotated. This is my cdr_csv.conf.xml (the dots ... is just so it didn't take up the entire screen when copy-paste): <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="example"/>   <param name="rotate-on-hup" value="true"/>  <param name="legs" value="a"/>  <param name="master-file-only" value="true"/> </settings> <templates>  <template name="example">"${caller_id_name}","${caller_id_number}",...    </templates> </configuration> 2012/2/13 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com>: > Hi Vitalie. > > I've already set rotate-on-hub to true. But I prefer running the > fs_cli command as I do not want the log files to be rotated as well. > And if I set the rotate-on-hub to false in the log-conf, no files get > rotated on hub. Not even the cdr records. > > On Mon, Feb 13, 2012 at 3:05 AM, Vitalie Colosov <vetali100(a)gmail.com> wrote: >> Ensure the following line is uncommented in cdr_csv.conf.xml >> >> <param name=”rotate-on-hup” value=”true”/> >> >> >> To rotate, execute from commandline: >> >> killall -HUP freeswitch >> >> You can add this command to cron so it will be executed automatically. >> >> >> >> >> 2012/2/12 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> >>> >>> I'm having FS running from the user freeswitch on CentOS 6. >>> >>> When calling (as the user root): >>> /usr/local/freeswitch/bin/fs_cli -x 'cdr_csv rotate' >/dev/null >>> nothing happens. The cdr-records does not get rotated. >>> >>> Can anyone help me on why this is happening and how I can rotate the >>> files as root? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >>

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