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FreeSWITCH Weekly Conference Call Starting Shortly, Please Start Calling In
by Michael Collins 16 Oct '09

16 Oct '09

The weekly conference call is about to start. Please call in. The agenda and calling instructions are here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16 Looking forward to hearing from everyone! -Michael

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Subject:, Re: FreeSWITCH-users Digest, Orien L. answer to Mike G.
by Orien Love 16 Oct '09

16 Oct '09

Mike, I copied the status screen from the freeswitch package in the web setup. I you need a console view, I will need instructions on how to console into pfsense, Since I have never done that. Orien L. Name Type Data State ================================================================================================= internal profile sip:mod_sofia@192.168.10.1:5060 RUNNING (0) external profile sip:mod_sofia@192.168.2.210:5080 RUNNING (0) 192.168.10.1 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias sofia status profile external ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.2.210 Ext-RTP-IP 192.168.2.210 SIP-IP 192.168.2.210 Ext-SIP-IP 192.168.2.210 URL sip:mod_sofia@192.168.2.210:5080 BIND-URL sip:mod_sofia@192.168.2.210:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= sofia status profile internal ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.10.1 Ext-RTP-IP 192.168.10.1 SIP-IP 192.168.10.1 Ext-SIP-IP 192.168.10.1 URL sip:mod_sofia@192.168.10.1:5060 BIND-URL sip:mod_sofia@192.168.10.1:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 6 FAILED-CALLS-IN 3 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: NjczMzQzNTVlYTJhYjIzYmFiZmFjZGYzNzNlMTVkMTc. User: 1001(a)192.168.10.1 Contact: "user" Agent: Pangolin v5.2, Build 09182009 Status: Registered(UDP)(unknown) EXP(2009-10-15 05:03:17) Host: orienhome2.local IP: 192.168.10.243 Port: 3789 Auth-User: 1001 Auth-Realm: 192.168.10.1 ================================================================================================= sofia status profile internal-ipv6 ================================================================================================= Name internal-ipv6 Domain Name N/A DBName sofia_reg_internal-ipv6 Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP ::1 SIP-IP ::1 URL sip:mod_sofia@[::1]:5060 BIND-URL sip:mod_sofia@[::1]:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= status UP 0 years, 0 days, 6 hours, 55 minutes, 32 seconds, 640 milliseconds, 638 microseconds 43 session(s) since startup 0 session(s) 0/30 1000 session(s) max > Subject: > Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L > From: > Michael Gende <mgende(a)gendesign.com> > Date: > Thu, 15 Oct 2009 09:36:55 -0500 > To: > freeswitch-users(a)lists.freeswitch.org > > To: > freeswitch-users(a)lists.freeswitch.org > > > Hey Orien, > > OK, let's have a look here. How about going to the console of your FS > and posting the output of the following commands: > > sofia status profile internal > sofia status profile external > > We'll take it from there. Your problems are just like what I ran into > initially, so I'm hopeful we get round them. > > Mike G. > > On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende(a)gendesign.com > <mailto:mgende@gendesign.com>> wrote: > > I'll get to you in the morning (where do you live?) and we'll see > what's what. > > > > On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien(a)tx.rr.com > <mailto:orien@tx.rr.com>> wrote: > > Mike, > Thank you for your quick response, this has been driving > me nuts. > > in my third attempt I tried to follow the mulit home example. > I did set > the phones to register to the local lan IP. actually had them > registering, but it did help it in any way still the same problem. > After my post I reinstalled the package and set the firewall > to allow > phones on the wan to connect, as per > http://doc.pfsense.org/index.php/FreeSWITCH#Firewall > put my computer with the soft phone on the wan network > (192,168.2.xxx) > had it registering and it had the same problem. so far all the > setups I > have tried are not working, the phones register, but cannot > dial/connect another phone, also does not go to voice mail, > maybe it > is in my dial plan. which is default. > > Shoot, I am not picky, I just want to get it to work. so any > suggestions > on how to accomplish this are greatly appreciated. > > My phones are currently a couple of polycom 601's, and a > Pangolin Softphone. > > Thanks > Orien L. > > > >

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Building Freeswitch with VPATH (src and obj directories are different)?
by Simon J Mudd 16 Oct '09

16 Oct '09

I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations. This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates. However this does not seem to work: [sjmudd@mad07 src]$ pwd /Users/sjmudd/src [sjmudd@mad07 src]$ ls -l | grep freeswitch drwxr-xr-x 3 sjmudd staff 102 13 Oct 07:52 freeswitch-obj drwxr-xr-x 56 sjmudd staff 1904 13 Oct 14:29 freeswitch-svn [sjmudd@mad07 src]$ cd freeswitch-obj [sjmudd@mad07 freeswitch-obj]$ ../freeswitch-svn/configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d checking for gawk... no checking for mawk... no checking for nawk... no checking for awk... awk checking whether make sets $(MAKE)... yes configure: error: source directory already configured; run "make distclean" there first [sjmudd@mad07 freeswitch-obj]$ cd ../freeswitch-svn [sjmudd@mad07 freeswitch-svn]$ make distclean Making clean in build Making clean in src Making clean in mod making clean mod_amr making clean mod_cdr_csv making clean mod_cluechoo making clean mod_commands making clean mod_conference making clean mod_console making clean mod_dialplan_asterisk making clean mod_dialplan_xml making clean mod_dptools making clean mod_enum making clean mod_esf making clean mod_event_socket making clean mod_expr making clean mod_fifo making clean mod_file_string making clean mod_fsv making clean mod_g723_1 making clean mod_g729 making clean mod_h26x making clean mod_iax making clean mod_ilbc making clean mod_limit making clean mod_local_stream making clean mod_logfile making clean mod_loopback making clean mod_lua making clean mod_native_file making clean mod_say_en making clean mod_siren making clean mod_sndfile making clean mod_sofia rm -f "./so_locations" making clean mod_speex making clean mod_spidermonkey making clean mod_spidermonkey_core_db making clean mod_spidermonkey_socket making clean mod_spidermonkey_teletone making clean mod_syslog making clean mod_tone_stream making clean mod_voicemail making clean mod_voipcodecs making clean mod_xml_cdr making clean mod_xml_rpc making clean mod_alsa making clean mod_amrwb making clean mod_celt making clean mod_cepstral making clean mod_dahdi_codec making clean mod_dialplan_directory making clean mod_dingaling making clean mod_easyroute making clean mod_erlang_event making clean mod_event_multicast making clean mod_fax making clean mod_flite WARNING mod_http is not a valid FreeSWITCH module dir, skipping it..... making clean mod_lcr making clean mod_ldap making clean mod_memcache making clean mod_nibblebill making clean mod_opal Package opal was not found in the pkg-config search path. Perhaps you should add the directory containing `opal.pc' to the PKG_CONFIG_PATH environment variable No package 'opal' found Package opal was not found in the pkg-config search path. Perhaps you should add the directory containing `opal.pc' to the PKG_CONFIG_PATH environment variable No package 'opal' found making clean mod_openzap rm -f "./so_locations" rm -f "./so_locations" rm -f "./so_locations" rm -f "./so_locations" rm -f "./so_locations" rm -f "./so_locations" rm -f "./so_locations" rm -f testtones testtones rm -f detect_tones detect_tones rm -f detect_dtmf detect_dtmf rm -f testisdn testisdn rm -f testpri testpri rm -f testr2 testr2 rm -f testboost testboost rm -f testanalog testanalog rm -f testapp testapp rm -f testcid testcid making clean mod_perl making clean mod_pocketsphinx making clean mod_portaudio making clean mod_python making clean mod_radius_cdr making clean mod_rss making clean mod_say_de making clean mod_say_es making clean mod_say_fr making clean mod_say_it making clean mod_say_nl making clean mod_say_ru making clean mod_say_zh making clean mod_shout making clean mod_skypiax making clean mod_snom making clean mod_soundtouch making clean mod_spidermonkey_odbc making clean mod_spy making clean mod_vmd making clean mod_xml_curl making clean mod_xml_ldap making clean mod_yaml Making clean in . Making clean in . rm -f freeswitch freeswitch rm -f fs_cli fs_cli rm -f fs_ivrd fs_ivrd rm -f "./so_locations" [sjmudd@mad07 freeswitch-svn]$ cd ../freeswitch-obj [sjmudd@mad07 freeswitch-obj]$ ../freeswitch-svn/configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d checking for gawk... no checking for mawk... no checking for nawk... no checking for awk... awk checking whether make sets $(MAKE)... yes configure: error: source directory already configured; run "make distclean" there first [sjmudd@mad07 freeswitch-obj]$ This is with [sjmudd@mad07 freeswitch-svn]$ svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 15138 Node Kind: directory Schedule: normal Last Changed Author: anthm Last Changed Rev: 15138 Last Changed Date: 2009-10-12 16:58:09 +0200 (Mon, 12 Oct 2009) Simon

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failed in bridging with SIPp
by Erwin Davis 16 Oct '09

16 Oct '09

Hi, I tried to originate a call from x-lite and hope to terminate to SIPp running in another machine (192.168.1.36:5069) through FS. My x-lite received service no available and the FS report NORMAL_TEMPOARY_FAILURE (http://pastebin.freeswitch.org/10720) My purpose for FS is to simply forward the call to the destination IP addr, which is 192.168.1.36. Did I miss anything in the configuration? Thanks, Below is my configuration, in dialplan/default.xml, <extension name="to-sipp"> <condition field="destination_number" expression="^6666$"> <action application="bridge" data="sofia/internal/profile1/$ 0(a)192.168.1.36"/> </condition> </extension> in sip-profile/profile1.xml <profile name="profile1"> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-ip" value="192.168.1.36"/> <param name="sip-port" value="5069"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> </profile>

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Alias user mapping
by Jonathan Barou 15 Oct '09

15 Oct '09

Hi everybody, I have a problem with "Alphanumeric to numeric user mapping" I have done like it's written here : Alphanumeric to numeric user mapping Say you want a user's id to be alphanumeric (like an email username), such as johnsmith(a)pbx.example.com. These users have alphanumeric usernames in their sip phone config, but you want to map them from their sip username to a numeric extension, and vice versa. As of version 1.0.4, Freeswitch makes this trivial to accomplish. A user's ID can be any alphanumeric string, and this can be simply tied to an extension number using the 'number-alias' property. This property creates an aliased directory entry that points to the alphanumeric user entry. *NOTE: When using this attribute, you must be careful not to create a directory collision by having another user whose ID is the same as another user's alias* Here is an example from the user directory: <user id="johnsmith" number-alias="1001">  </user> So when a user dials extension number 1001, your dialplan can use the 'user_data' function to look up the ID attribute associated with that number alias. In the default dialplan, the 'Local Extension' section can be made to work with a small change to the 'bridge' line: <action application="bridge" data="user/${user_data(${dialed_extension}@${domain_name} attr id)}@${domain_name}"/> *NOTE: Using this user_data function in combination with mod_xml_curl will generate an additional request each time the user_data function is called. Note that it is already called once in the Local Extension section to determine the callgroup. Beware of performance implications of this with high-volume systems.* But when I want to call my alias-number, FS says "No Route, Abording" My version of FreeSWITCH is the 1.0.4pre9. Do you have any ideas ? Did I forget something to do? Thanks -- Jonathan BAROU SQLI LYON - CRCI jbarou(a)sqli.com lyon.crci(a)sqli.com

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lua script causing FreeSwitch to crash?
by Lars Zeb 15 Oct '09

15 Oct '09

http://pastebin.freeswitch.org/10686 I am trying to use a lua script for inbound calls. The caller hears a busy signal. After this call fails to go through, FreeSwitch is no longer in memory - it does not appear in 'ps -ef' output. I don't have a clue what I might be doing incorrectly in the script in order to cause this to happen. There is no indication in the log that FreeSwitch has terminated. But it is not resident in memory. I am using v15134. I had been using this same script without incident as late as v14929. Until I can resolve the issue, I have removed the call to lua from the dialplan and done all I could in it instead. I have included the dialplan, the script and the cli log in the pastebin. I would appreciate any help or suggestions. Thanks, Lars

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apr_queue
by srinivasula reddy 15 Oct '09

15 Oct '09

Hi all, does any know about How apr_queue is maintaing and retriving all registered and all stuff.... -- Srinivasula Reddy K

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WG: Changing display text on phones (sip_from_user)
by Weigel, Stefan 15 Oct '09

15 Oct '09

Hi, just found the answer: just use "effective_caller_id_name" and it work's :) Cheers Stefan Stefan Weigel Advanced IT-Professional Tel.: +49 89 2000 48 975 Fax: +49 89 2000 48 566 eMail: Stefan.Weigel(a)allianz-warranty.com<mailto:Stefan.Weigel@allianz-warranty.com> Allianz Automotive Services GmbH Einsteinring 28 85609 Aschheim Germany http://www.allianz-warranty.com Geschäftsführung: Andreas Rösing, Horst Ziegler Amtsgericht München, HRB 175682 Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720 Von: Weigel, Stefan Gesendet: Donnerstag, 15. Oktober 2009 17:16 An: 'freeswitch-users(a)lists.freeswitch.org' Betreff: Changing display text on phones (sip_from_user) Hi list, I'm playing around with this wonderful, mighty solution. Setup so far: I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones): <condition field="${sip_to_user}" expression="^0.*974$" break="never"> <action application="set" data="sip_from_user='IT_Hotline'"/> ... My telephone rings but it's always showing the original dialer number. I also tried the variables username caller_id_name sip_contact_user in the application info log it's showing me: Caller-Username: [01732898376] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [01732898376] Caller-Caller-ID-Number: [01732898376] [..] variable_sip_from_user: ['IT_Hotline'] variable_sip_from_user_stripped: ['IT_Hotline'] variable_sip_contact_user: ['IT_Hotline'] Is there any chance to rewrite only the part that's getting displayed on the phone (e.g. From: IT_Hotline sip:123456@sip.qsc.de) ? Thanks and best regards Stefan Stefan Weigel Advanced IT-Professional Tel.: +49 89 2000 48 975 Fax: +49 89 2000 48 566 eMail: Stefan.Weigel(a)allianz-warranty.com<mailto:Stefan.Weigel@allianz-warranty.com> Allianz Automotive Services GmbH Einsteinring 28 85609 Aschheim Germany http://www.allianz-warranty.com Geschäftsführung: Andreas Rösing, Horst Ziegler Amtsgericht München, HRB 175682 Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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Changing display text on phones (sip_from_user)
by Weigel, Stefan 15 Oct '09

15 Oct '09

Hi list, I'm playing around with this wonderful, mighty solution. Setup so far: I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones): <condition field="${sip_to_user}" expression="^0.*974$" break="never"> <action application="set" data="sip_from_user='IT_Hotline'"/> ... My telephone rings but it's always showing the original dialer number. I also tried the variables username caller_id_name sip_contact_user in the application info log it's showing me: Caller-Username: [01732898376] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [01732898376] Caller-Caller-ID-Number: [01732898376] [..] variable_sip_from_user: ['IT_Hotline'] variable_sip_from_user_stripped: ['IT_Hotline'] variable_sip_contact_user: ['IT_Hotline'] Is there any chance to rewrite only the part that's getting displayed on the phone (e.g. From: IT_Hotline sip:123456@sip.qsc.de) ? Thanks and best regards Stefan Stefan Weigel Advanced IT-Professional Tel.: +49 89 2000 48 975 Fax: +49 89 2000 48 566 eMail: Stefan.Weigel(a)allianz-warranty.com<mailto:Stefan.Weigel@allianz-warranty.com> Allianz Automotive Services GmbH Einsteinring 28 85609 Aschheim Germany http://www.allianz-warranty.com Geschäftsführung: Andreas Rösing, Horst Ziegler Amtsgericht München, HRB 175682 Für Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L
by Orien Love 15 Oct '09

15 Oct '09

Mike, Thank you for your quick response, this has been driving me nuts. in my third attempt I tried to follow the mulit home example. I did set the phones to register to the local lan IP. actually had them registering, but it did help it in any way still the same problem. After my post I reinstalled the package and set the firewall to allow phones on the wan to connect, as per http://doc.pfsense.org/index.php/FreeSWITCH#Firewall put my computer with the soft phone on the wan network (192,168.2.xxx) had it registering and it had the same problem. so far all the setups I have tried are not working, the phones register, but cannot dial/connect another phone, also does not go to voice mail, maybe it is in my dial plan. which is default. Shoot, I am not picky, I just want to get it to work. so any suggestions on how to accomplish this are greatly appreciated. My phones are currently a couple of polycom 601's, and a Pangolin Softphone. Thanks Orien L. > > ------------------------------------------------------------------------ > > Subject: > Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129 > From: > Michael Gende <mgende(a)gendesign.com> > Date: > Wed, 14 Oct 2009 21:57:57 -0500 > To: > freeswitch-users(a)lists.freeswitch.org > > To: > freeswitch-users(a)lists.freeswitch.org > > > Hey Orien, > > All of these guys are more experienced than I, but I'll try to throw > you a line. > > I'll have to check a thing or two out at work, but here's an > observation or two (I'll post again tomorrow morning after I get in). > > The mult-homed example is set to work in a very specifc way: A > dual-homed box where the WAN is used for the WWW connection, and then > using the LAN side for phones. > > To work like the example, you need to have your phones on the LAN > network. Then, configure your internal.xml file on FS to use the LAN > port, not the WAN. Next, set your phones to register to that LAN > address. > > One thing I prefer to do on pfsense, is at the FS box console (or ssh > to the box), kill the "pfsense FS" running the background, and run FS > in the foreground. Then you can "reloadxml" instead of rebooting when > XML changes are made (if I understand your post). > > As I said, let me check a thing or two out, and I'll ask to see some > configuration files. I'm sure we can get you going. > > BTW: What kind of phones are you using? > > Regards, > > Mike G. > > > On Wed, Oct 14, 2009 at 6:31 PM, Orien Love <orien(a)tx.rr.com > <mailto:orien@tx.rr.com>> wrote: > > Hello, a few weeks ago I asked about a setup, and about using the > freeswitch package on pfSense. Since then I have received most of my > hardware and started setting up freeswitch. > > First my setup > WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w > freeswtich <-- Test (192.168.3.xxx) --> phones > > pfsense w freeswtich Port Addresses > WAN 192.168.2.210 > LAN 192.168.3.1 > > this is setup to simulate a pfsense box hooked up to an external > network. > > > here are the things I have done. > > 1st attempt > #1 set my WAN IP port to fixed in pfsense (192.168.2.210) > #2 set my soft phone to register to 192.168.2.210 (The phones > registered > and showed up in the status section) > #3 Tried to make a call to extension 1004, the soft phone tells me > that > it is "Temporally Unavailable 480). > > 2nd attempt > #1 set my WAN IP port to fixed in pfsense (192.168.2.210) > #2 deleted the lan.xml profile > #3 rebooted freeswitch. > #4 set my soft phone to register to 192.168.2.210 (The phones > registered > and showed up in the status section) > #5 Tried to make a call to extension 1004, the soft phone tells me > that > it is "Temporally Unavailable 480). > > 3rd attempt > #1 set my WAN IP port to fixed in pfsense (192.168.2.210) > #2 edited the internal.xml file as described in the multi-home > tutorial > (192.168.3.1 is my phone lan address) > <param name="rtp-ip" value="192.168.3.1"/> > <param name="sip-ip" value="192.168.3.1"/> > <param name="ext-rtp-ip" value="192.168.3.1"/> > <param name="ext-sip-ip" value="192.168.3.1"/> > #3 rebooted freeswitch. > #4 set my soft phone to register to 192.168.2.210 (The phone would not > register) > #5 set my soft phone to register to 192.168.3.1 (The phone registered > and showed up in the status section) > #6 Tried to make a call to extension 1004, the soft phone tells me > that > it is "Temporally Unavailable 480). > #7 edited the file /usr/local/freeswitch/conf/directory/default.xml > <domain name="192.168.3.1"> > #8 rebooted freeswitch. > #9 set my soft phone to register to 192.168.2.210 (The phone would not > register) > #10 set my soft phone to register to 192.168.3.1 (The phone would not > register got a forbidden error) > #11 did some reading and found a place in the vars tab to change > the domain > #12 edited the file in the vars tab > <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/> > #13 rebooted freeswitch. > #14 set my soft phone to register to 192.168.2.210 (The phone > would not > register) > #15 set my soft phone to register to 192.168.3.1 (The phone registered > and showed up in the status section) > #16 Tried to make a call to extension 1004, the soft phone tells > me that > it is "Temporally Unavailable 480). > > here is part of the log file: > 2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan > 2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker > thread > 2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan > 2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199 > Rejected > by acl "domains". Falling back to Digest auth. > 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199 > Rejected > by acl "domains". Falling back to Digest auth. > 2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3] > 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode > based on > rfc1918.auto > 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > entering state [received][100] > 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP: > v=0 > o=- 78902911 78902911 IN IP4 192.168.189.1 > s=http://www.portsip.com > c=IN IP4 192.168.189.1 > t=0 0 > m=audio 21328 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > m=video 40216 RTP/AVP 34 125 > a=rtpmap:34 H263/90000 > a=fmtp:34 CIF=2 > a=rtpmap:125 H264/90000 > a=fmtp:125 profile-level-id=42e015 > > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec > Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec > Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec > Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec > Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > PCMU/8000 20 ms 160 samples > 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf > payload to 101 > 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State Change CS_NEW -> CS_INIT > 2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send > signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [BREAK] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Running State Change CS_INIT > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State INIT > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > SOFIA INIT > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State Change CS_INIT -> CS_ROUTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send > signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [BREAK] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State INIT going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Running State Change CS_ROUTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State ROUTING > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > SOFIA ROUTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > Standard ROUTING > 2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing > 1001->1004 in context public > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> parsing [public->unloop] > continue=false > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> parsing > [public->outside_call] continue=true > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> Absolute Condition > [outside_call] > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> Action set(outside_call=true) > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> parsing [public->call_debug] > continue=true > Dialplan: sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> Regex (FAIL) [call_debug] > ${call_debug}(false) =~ /^true$/ break=never > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State Change CS_ROUTING -> CS_EXECUTE > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send > signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [BREAK] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State ROUTING going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Running State Change CS_EXECUTE > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State EXECUTE > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > SOFIA EXECUTE > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > Standard EXECUTE > EXECUTE sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> set(outside_call=true) > 2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> SET > [outside_call]=[true] > 2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179 > Hangup sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> [CS_EXECUTE] [NORMAL_CLEARING] > 2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [KILL] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send > signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [BREAK] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State EXECUTE going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Running State Change CS_HANGUP > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State HANGUP > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > hanging up, cause: NORMAL_CLEARING > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE > with: 480 > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > Standard HANGUP, cause: NORMAL_CLEARING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State HANGUP going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State Change CS_HANGUP -> CS_REPORTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send > signal > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> [BREAK] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Running State Change CS_REPORTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State REPORTING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > Standard REPORTING, cause: NORMAL_CLEARING > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State REPORTING going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State Change CS_REPORTING -> CS_DESTROY > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > Locked, Waiting on external entities > 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085 > Session 2 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) Ended > 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/internal/1001(a)192.168.2.210 > <mailto:1001@192.168.2.210> [CS_DESTROY] > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State DESTROY > 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > SOFIA DESTROY > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210> > Standard DESTROY > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State DESTROY going to sleep > 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 > (sofia/internal/1001(a)192.168.2.210 <mailto:1001@192.168.2.210>) > State DESTROY going to sleep > > I am at a loss, any help would be welcome. > > Orien Love > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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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