- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch and Exchange 2007
by Drago Totev 23 Jun '09

23 Jun '09

Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: "MS Exchange does not understand 'From' fields with domain name in the address (example: field containing <sip:100@3CXPhoneSystem> address).The 'From' field's address to be in the format '<sip:DN@IP:Port>'; otherwise, MS Exchange drops the call." I don't know if this is the only problem. However, I see exactly this behavior: "PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established." After "302 (Moved Temporarily", FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago

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How to enable compact SIP headers in mod_sofia
by Muhammad Shahzad 23 Jun '09

23 Jun '09

Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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How to change outbound_caller_name=FreeSWITCH and number=000000000
by Edmar Cruz 23 Jun '09

23 Jun '09

When I calling an outbound extension it appears: name is FreeSWITCH and number is 000000000 How can i change it depends on the user who is calling? Sample 1001->64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-n… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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How to originate gtalk calls
by Jingwei Yang 23 Jun '09

23 Jun '09

Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch> *originate dingaling/gmail.com/userAAA@gmail.com &echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. Thanks!

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rpm build issues on CentOS 5 with trunk
by Joseph L. Casale 22 Jun '09

22 Jun '09

I attempted to build rpm's from the included spec file using a non-root user build environment. The steps I used are as follows: 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as expected by the spec: svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4 tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ 3. Copy spec to SPECS directory: 4. Pull in Source (Source0 doesn't exist yet, I make it above): for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done 5. Check spec in svn copy for deps: yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' freeswitch.spec) 6.Build rpm: rpmbuild -ba freeswitch.spec After some time, near the end I see various issues like the following: making install mod_speex installing mod_speex.so quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la' has not been installed in `/opt/freeswitch/lib' It also seems to download everything it would normally again, then fails with several errors like the following: RPM build errors: File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so* Installed (but unpackaged) file(s) found: /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml . . . After which no rpm's are built? Anyone know what tricks are still needed with the spec from svn? Thanks! jlc

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Limit length of call with mod_limit?
by Lon Baker 22 Jun '09

22 Jun '09

Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon

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Re: [Freeswitch-users] Transmit fax locally for test
by Tim B 22 Jun '09

22 Jun '09

8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000(a)192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian(a)freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

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Re: [Freeswitch-users] Transmit fax locally for test
by Tim B 22 Jun '09

22 Jun '09

Just wanted to follow-up again. Is this the proper or best way to configure this? See below... Tim From: timb0311(a)hotmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: (Found Fix) Transmit fax locally for test Date: Fri, 19 Jun 2009 23:06:10 -0400 Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: <list name="domains" default="deny"> <node type="allow" cidr="192.168.10.35/32"/> <node type="allow" domain="$${domain}"/> </list> So now this statement works for local fax testing: originate sofia/default/8000(a)192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins <msc(a)freeswitch.org> > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503(a)mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000(a)x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B <timb0311(a)hotmail.com> > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users(a)lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > Bing™ brings you maps, menus, and reviews organized in one place. Try it now. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing™ now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_… bing_1x1

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How to change sound-path when switch language
by Dome Charoenyost 22 Jun '09

22 Jun '09

Dear sir, i create mod_say_th for Thai language. i found some problem about sound-path. I have config th.xml in conf/lang/th/ <language name="en" sound-path="$${base_dir}/sounds/en" tts-engine="cepstral" tts-voice="callie"> ... when i try <action application="say" data="th number pronounced 20230021"/> Freeswitch still looking sounf file in /sounds/en/us/callie (en sound-path) Someone help me please Dome C.

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Question about bridging calls to a specific URI via a specific profile
by Darren Schreiber 22 Jun '09

22 Jun '09

Hello, I was wondering, I am bridging a call to a specific URI as follows: <action application="bridge" data="sofia/internal/3032223232(a)3.55.66.180;fs_nat=yes;fs_path=sip%3As%403.5 5.66.180%3A7812"> EXECUTE sofia/internal/+17209460000@ <mailto:sofia/internal/+17209460000@2.3.4.5> 2.3.4.5 bridge(sofia/internal/3032223232(a)3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223232@3.55.66.180;fs_nat=yes;fs_path=sip@3As%403.55.66.180%3A7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/URI@domain that FS will use the specified SIP profile to try and connect a call to the URI@domain specified. Since the full URI@domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:XXX@domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber

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