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Newbie trouble with default config extension 9198...
by Francisco Javier Cintrón Olguín 17 Oct '11

17 Oct '11

Hi there. :) After spending hours trying to register Xlite 4.1 in freeswitch 1.0.6 both installed in my Windows 7 pro rig I gave up. Now I installed Freeswtich 1.0.6 in a centos 5.7 x86 virtual machine with the help of freeswitch book. Xlite is registering fine but now the problem is this: When I dial 9198 extension Xlite tells me "Failed to establish call" and This log messages are written in freeswitch console: freeswitch@pbx> 2011-10-12 09:01:17.469489 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)10.10.100.35[e3f7697a-74ff-46a6-a3ac-c5b95f58d103] 2011-10-12 09:01:17.474982 [INFO] mod_dialplan_xml.c:418 Processing 1000->9198 in context default 2011-10-12 09:01:17.480554 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000(a)10.10.100.35 to enum[9198@default] 2011-10-12 09:01:17.497178 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-10-12 09:01:17.497178 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000(a)10.10.100.35 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-10-12 09:01:17.502715 [NOTICE] switch_core_session.c:1182 Session 6 (sofia/internal/1000(a)10.10.100.35) Ended 2011-10-12 09:01:17.504038 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000(a)10.10.100.35 [CS_DESTROY] It looks that 9198 extension is not defined. This is the default configuration I haven't change anything. If it helps extension 5000 is working. What am I doing wrong? Thanks in advance for your help and have a good day.

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Re: [Freeswitch-users] Freeswitch (Windows) 64-bit - call answering issues
by Network Information Centre 17 Oct '11

17 Oct '11

Oddly, testing this from another external sip (android) phone on my switch to another external phone - WORKS. I suspect this may be either a) an issue with my softphone client - although it worked prior) Or b) something on my internal network is not nating or routing traffic to/from my softphone (as my switch is now externally hosted). Either way, It appears to be working outside my test network. ------Original Message------ From: Jeff Lenk Sender: freeswitch-users-bounces(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org ReplyTo: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch (Windows) 64-bit - call answering issues Sent: Oct 17, 2011 12:21 PM Did you copy your config files from the other machine? Did you modify the default configuration to set local ip addresses to deal with NAT perhaps? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Windows-64-bit-cal… Sent from the freeswitch-users mailing list archive at Nabble.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 Regards, Network Informatiom Centre e-City Solutions Inc. nic(a)itecity.com

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mod_blacklist functionality change input
by curriegrad2004 17 Oct '11

17 Oct '11

Hey all, Currently I am looking for community input on how mod_blacklist should be changed. Right now I find that when the blacklist dump api command is executed in the console, it only dumps the numbers it has on the hash table. That may not be much of a concern for most users using this, but what if you have to remove a few numbers from the blacklist? Currently from what I can see on mod_blacklist is it only dumps the numbers contained in the hash table and it won't make any attempts to clear out the old numbers that are already stored in the blacklist file. My proposed change would be to make blacklist dump to flush out the contents first and THEN write out the contents on the hash table to that file as specified on the xml config, this change has been already pushed out on my github repo as commented out code as I have not even bothered to make any attempts to test this functionality change... yet... I might test this when I get home tonight from school, but that depends if I do have time on hand to even test this out. Regards, curriegrad2004

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Freeswitch (Windows) 64-bit - call answering issues
by Network Information Centre 17 Oct '11

17 Oct '11

Hello, I am rather new to Freeswitch, but have successfully installed and ran the 32-bit windows version. I have now upgraded to the windows 64-bit, on a larger server (pretty clean re-install). I can make and receive calls (through multiple gateways). However, my issue is that the call is never actually answered (but Freeswitch says it is). Although, I believe I may actually have two different issues. 5555551212 = Freeswitch line/extension 15554445555 = Remote line Calls are being made between a cellular phone and a softphone (connected to Freeswitch) 1) If placing a call, the remote side answers (and Freeswitch also reports this), but the calling side just keeps ringing. 2011-10-16 10:41:44.146140 [NOTICE] switch_channel.c:911 New Channel sofia/internal/5555551212@sip.XXXXX.com:5060 [4f422e4e-2513-46a2-8416-cd1a771a231a] 2011-10-16 10:41:44.193015 [INFO] mod_dialplan_xml.c:336 Processing Don Hunter <5555551212>->15554445555 in context default 2011-10-16 10:41:44.193015 [NOTICE] switch_channel.c:911 New Channel sofia/external/15554445555 [54540bc5-e9aa-4485-ac6a-88ea43552dde] 2011-10-16 10:41:49.166648 [NOTICE] sofia_glue.c:3868 Pre-Answer sofia/external/15554445555! 2011-10-16 10:41:49.166648 [INFO] switch_ivr_originate.c:3198 Sending early media 2011-10-16 10:41:49.166648 [NOTICE] mod_sofia.c:2464 Pre-Answer sofia/internal/5555551212@sip.XXXXX.com:5060! 2011-10-16 10:41:51.018210 [NOTICE] sofia.c:5780 Channel [sofia/external/15554445555] has been answered 2011-10-16 10:41:51.018210 [NOTICE] switch_ivr_bridge.c:417 Channel [sofia/internal/5555551212@sip.XXXXX.com:5060] has been answered 2011-10-16 10:41:59.188132 [NOTICE] sofia.c:560 Hangup sofia/external/15554445555 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-10-16 10:41:59.209617 [NOTICE] switch_ivr_bridge.c:1355 Hangup sofia/internal/5555551212@sip.XXXXX.com:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2011-10-16 10:41:59.209617 [NOTICE] switch_core_session.c:1367 Session 6 (sofia/external/15554445555) Ended 2011-10-16 10:41:59.209617 [NOTICE] switch_core_session.c:1369 Close Channel sofia/external/15554445555 [CS_DESTROY] 2011-10-16 10:41:59.495750 [NOTICE] switch_core_session.c:1367 Session 5 (sofia/internal/5555551212@sip.XXXXX.com:5060) Ended 2011-10-16 10:41:59.495750 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/5555551212@sip.XXXXX.com:5060 [CS_DESTROY] As you can see by the [2011-10-16 10:41:51.018210] lines, the call is reporting as being answered, but is not actually picked up. 2) When receiving a call, the calling side rings and eventually goes to voicemail. But the called side also rings, but even when answered (phone set shows answered) both sides continue to ring. 2011-10-16 10:58:52.552476 [NOTICE] switch_channel.c:911 New Channel sofia/internal/15554445555(a)46.19.209.14 [ff76c548-816f-4313-8b15-dbca896e1264] 2011-10-16 10:58:52.589585 [INFO] mod_dialplan_xml.c:336 Processing 4167869038 <15554445555>->5555551212 in context public 2011-10-16 10:58:52.589585 [NOTICE] switch_ivr.c:1699 Transfer sofia/internal/15554445555(a)46.19.209.14 to XML[5555551212@default] 2011-10-16 10:58:52.589585 [INFO] mod_dialplan_xml.c:336 Processing 4167869038 <15554445555>->5555551212 in context default 2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *1 execute_extension::dx XMLfeatures 2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *2 record_session::C:\Programecordings/archive/2011/Oct/16/ff76c548-816f-4313-8 b15-dbca896e1264.wav 2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *3 execute_extension::cf XMLfeatures 2011-10-16 10:58:52.589585 [NOTICE] switch_channel.c:911 New Channel sofia/internal/sip:5555551212@69.196.144.178:33677 [36da2d38-0b99-47ae-9f1b-5fa7785159ee] 2011-10-16 10:58:52.741929 [NOTICE] sofia.c:5235 Ring-Ready sofia/internal/sip:5555551212@69.196.144.178:33677! 2011-10-16 10:58:52.741929 [INFO] switch_ivr_originate.c:1113 Sending early media 2011-10-16 10:58:52.741929 [NOTICE] mod_sofia.c:2464 Pre-Answer sofia/internal/15554445555(a)46.19.209.14! 2011-10-16 10:59:22.008531 [NOTICE] switch_ivr_originate.c:3165 Hangup sofia/internal/sip:5555551212@69.196.144.178:33677 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-10-16 10:59:22.008531 [NOTICE] switch_ivr_originate.c:2457 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2011-10-16 10:59:22.008531 [INFO] mod_dptools.c:2810 Originate Failed. Cause: NO_ANSWER 2011-10-16 10:59:22.008531 [NOTICE] mod_dptools.c:1043 Channel [sofia/internal/15554445555(a)46.19.209.14] has been answered 2011-10-16 10:59:22.008531 [NOTICE] switch_core_session.c:1367 Session 10 (sofia/internal/sip:5555551212@69.196.144.178:33677) Ended 2011-10-16 10:59:22.008531 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:5555551212@69.196.144.178:33677 [CS_DESTROY] 2011-10-16 10:59:32.526109 [NOTICE] sofia.c:560 Hangup sofia/internal/15554445555(a)46.19.209.14 [CS_EXECUTE] [NORMAL_CLEARING] 2011-10-16 10:59:32.867906 [NOTICE] switch_core_session.c:1367 Session 9 (sofia/internal/15554445555(a)46.19.209.14) Ended 2011-10-16 10:59:32.867906 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/15554445555(a)46.19.209.14 [CS_DESTROY] I have compared my configs against the version that works, but cannot see any differences. I know I am missing something small and stupid. Any assistance would be appreciated (even a pointer as to where to look)... as I am not sure where this is actually handled and in which XML(s). Regards, Don Hunter e-City Solutions Inc. email: don.hunter(a)44ecity.com CA/US Toll-Free: +1-855-IT-ECITY, ext 200 UK National: +44 870-820-0286 iNum (inum.net) +883 5100-0827-0154 ITeCity.com | FoundU.com | StoreLocatorService.com | go-Postal.ca | getZiped.com | FraudCloud.com WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com | VOIPMuch.com

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How do I debugging Client Auth?
by Jacek 17 Oct '11

17 Oct '11

Hi I have FreeSWITCH on Debian Squeeze. I have 2 questions: 1: How to configure FreeSWITCH, any list handling user checked the sql database (mysql) instead of the xml files in the folder conf / directory. Or, how to use external script for authentication (eg, use PAM, Radius, or otherwise authorized client)? 2: How to debug (verbose) a problem with user authentication (the client auth)? for example, the program (Twinkle) reports the error: +++ 17-10-2011 16:23:18.998049 INFO SIP ::process_sip_msg Received from: udp:83.230.27.214:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 83.230.27.214:5063;rport=5063;branch=z9hG4bKclobopii From: "user" <sip:user@domain.tld>;tag=nmehl To: "user" <sip:user@domain.tld>;tag=vQ0ma0yHt7UmH Call-ID: wiqutrgrresdddt(a)domain.tld CSeq: 381 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fdb4031 2011-10-14 14-13-16 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 and no trace of FreeSWITCH logs logon failure, or other cause of the problem. Console says only: soa_set_params(static::0x818deb0, ...) called tport_tsend(0x8172090) tpn = UDP/83.230.27.214:5063 tport_resolve addrinfo = 83.230.27.214:5063 tport_by_addrinfo(0x8172090): not found by name UDP/83.230.27.214:5063 tport_vsend returned 510 nta: sent 403 Forbidden for REGISTER (651) nta_leg_destroy((nil)) Where UDP/83.230.27.214:5063 - client addres. My Freeswitch comes from subversioon: version: Freeswitch (1.0.7) config.log: ..... uname -m = i686 uname -r = 2.6.39-bpo.2-686-pae uname -s = Linux uname -v = #1 SMP Thu Aug 4 11:02:22 UTC 2011 .... Cheers ;)

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Snom 370 Presence Function
by afshin afzali 17 Oct '11

17 Oct '11

Hi Guys, I'll appreciate if I know someone has been experienced snom 370 presence function. As I trace sip messages, it shows that snom does not subscribe for itself and just tries to publish its state which replies 404 by freeswitch ! I don't know it is the snom or freeswitch protocol issue. BEST, -- afshin

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Hold and BLF - disable flashing or pickup for held call?
by Alex Crow 17 Oct '11

17 Oct '11

Hi, Anyone know if there is a way to change notify behaviour in FS to avoid the below confusion (eg on Snom, you can intercept a call that someone you monitor has held), up to and including disabling NOTIFY for a held endpoint but not for a ringing one? Thanks, Alex ******* Hi, I have some snom 370s, and I noticed that when a monitored extension has held a call then the corresponding BLF lamp flashes exactly as if the monitored extension is ringing, and it is then possible to steal the call from the "holder" by pressing the button. Ideally I'd like neither of these things to happen. Is it possible to disable one or both for a held call? I have a polycom IP 650 with productivity licence and it shows the status of the holding extension as ringing (with the musical notes animation on the status display - not great) but at least it doesn't steal the call. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Re: [Freeswitch-users] [asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls &etc
by Alec Taylor 16 Oct '11

16 Oct '11

Thanks, looks like: sipp.sourceforge.net supports scripting... Are there sample University Curricula for teaching VOIP with Asterisk or FreeSwitch? On Sun, Oct 16, 2011 at 5:26 AM, Daniel Tryba <daniel(a)tryba.nl> wrote: > On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Taylor wrote: >> If asterisk or freeswitch would be taught in a classroom environment, >> is there someway to emulate and script emulation of users calling >> in/receiving calls, transferring calls &etc? > > The Asterisk Manager Interface (AMI) and callfiles to/from Echo(), > voicemail or IVRs! But much easier is to have multiple (soft)phones > available to 1 student. > > -- > > Daniel Tryba > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >

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Newbie problem: IVR at extension 5000 talking too fast...
by Francisco Javier Cintrón Olguín 15 Oct '11

15 Oct '11

Hi there. Well I am having some problems with default extension 5000, this is a barely new freeswitch installation. Actually I haven't changed anything from the default configuration. This morning I realized that the virtual machine which I used to install freeswitch had wrong time info, it had UTC time but my zone is America/Mexico_City After I adjusted it, I dialed 5000 extension and the IVR's voice was talking faster than before I changed time info. I am sure time's difference is the problem, but ¿why? And of course ¿How can I solve this problem? Thanks ind advance for your kind help. Have a good day.

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Question about sound files...
by Francisco Javier Cintrón Olguín 15 Oct '11

15 Oct '11

Are there spanish sound files for freeswitch? I am using version 1.0.6 and I am realizing that there are only russian and english sound files. Thanks in advance for your help. Have a good day.

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