April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problems with initial setup - basic nat
by Jay Reeder 26 Apr '08

26 Apr '08

We're setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I've wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. I've been through the wiki and reviewed list archives but I'm missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall's external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where I'm stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won't accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. I'm sure it's something obvious but what am I missing? Thanks, Jay

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make install error
by Angelito Manansala 26 Apr '08

26 Apr '08

Hi Guys, I've been playing with FS a week ago, I'm trying to install wikipbx following their installation instruction wiki now I have a error on make install command, below are some portions of the logs. mod_cepstral.c:379: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'cepstral_numeric_param_tts': mod_cepstral.c:389: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'cepstral_float_param_tts': mod_cepstral.c:401: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'mod_cepstral_load': mod_cepstral.c:410: error: 'engine' undeclared (first use in this function) make[5]: *** [mod_cepstral.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_cepstral-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -- Regards, Lito

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A small question about Freeswitch security
by Krzysiek 26 Apr '08

26 Apr '08

Hi once again :) I have a small question, because I don't know whether I understand Freeswitch security things correctly... If uncomment this line in the configs: <param name="Inbound-no-media" value="true"> and enable SRTP encryption (key via SDES or MIKEY) in phones, will I have to have support for those SDES or MIKEY in Freeswitch ? I thnik that sdp messages within sip will be passed through Freeswitch unchanged and i don't have to have support for those in Freeswitch and the call will be secure. I'm writing about private network, where calls are routed only between endpoionts in this private network. It is a theoretical question and I know that Freeswitch doesn't have MIKEY support. sorry for my english, I hope you understand me. Chris

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Hung Phone
by Luis Jimenez 26 Apr '08

26 Apr '08

Hi, i have this in default.xml <extension name="*99"> <condition field="destination_number" expression="^\*99$" break="always"> <action application="set" data="myvar=1004"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/default/${myvar}%10.0.0.100"/> </condition> </extension> when i call, it works, but when i hangup the extension 1004 (callee), my caller phone stays connected until i hagup manually this one. if i hangup the caller first the calee hangup fine. Luis jimenez

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Syntax for SIP2Jingle Call in Dialplan
by Chris Chen 25 Apr '08

25 Apr '08

I have setup one jabber server running along the Freeswitch on the centos 5, and setup the gtalk in public context as the jabber server component on port 5347 , and established the federation with googletalk network. Now I can call from gtalk client to my sip account on freeswitch without problem by the following dialplan: <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/default/$1%192.168.1.253"/> </condition> </extension> where 192.168.1.253 is the ip address of freeswitch. but I am having issues with the sip2jingle call. in the default context dialplan, I can do this <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^1500$"> <action application="bridge" data="dingaling/ jabber.mydomain.com/mybuddy@gmail.com"/> </condition> </extension> where the jabber.mydomain.com is the component profile name I defined for the gtalk server component. but if I modify the dialplan to do more general sip2jingle call as below: <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/ jabber.mydomain.com/$1@gmail.com"/> </condition> </extension> or <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@ ${sip_fromhost}/$1(a)gmail.com"/> </condition> </extension> I tested with jingle+mybuddy.lastname(a)gmail.com<jingle%2Bmybuddy.lastname(a)gmail.com>from my extension 1000 on soft phone Bria 2.2 on XP, I got the error " 1000(a)gmail.com not found" Could you guys share the correct syntax for doing SIP to Jingle calls? Thanks, Chris

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Shall we revive the ailing mod_cdr?
by Yossi Neiman 25 Apr '08

25 Apr '08

Hi folks, It's been quite a while. Some of you might know of the mod_cdr that was fully functional up until around sometime after July 2007. Due to personal issues, I did not have time to maintain it. I am now in a position to revive it. I will be making some changes to how it is designed, and removing some of the serialization. Currently in FreeSWITCH we have the following CDR logging formats at hand (separate of mod_cdr): Individual XML flat files, comma separate values, and curl HTTP handling. As such there is no need to include these in mod_cdr anymore, but I am looking for what backends everybody would be interested in. I will be starting off with the ODBC backend. Please note that database-specific backends are possible, but there is a possibility they would not be available for inclusion in trunk. Another format that I thought just for the fun of it would maybe to shoot out CDRs as e-mails (donno if that's at all useful...). Give me your thoughts and ideas. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com

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mod_commands linker error on Visual studio 2008
by B Karthik 25 Apr '08

25 Apr '08

Hi All, mod_commands does not seem to link on VS 2008. It complains about two unresolved functions - switch_console_set_alias, switch_console_set_complete. I understand that this is because libedit library used by freeswitch is not compiled on windows. It was solved by adding the two functions in switch_console.c. Both the functions returns SWITCH_STATUS_FALSE. Is there a better way to do this? Thanks in advance. Warm Regards, B Karthik.

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Massively Scalable Voicemail-to-Email Farm
by Guy Fawkes 24 Apr '08

24 Apr '08

FreeSWITCH Users, We are looking to set-up a simple, inbound-only, massively scalable voicemail platform running on Debian or FreeBSD. The goals are as follows: Set-up one or more SIP peers for inbound traffic. All calls are inbound; no outbound dialing is allowed. Only options are to dial-in leave a message or enter your PIN and record a new custom greeting. No messages are stored on the server and all are emailed out as mp3 attachments. The database should contain the mappings for the telephone number, user pin, and email address target. Creation and management of the user data such as the email address and PIN will be handled by an external process. Obviously, the custom greeting identifer will have to get linked to the telephone number data somehow so the system plays the correct custom greeting for any given dialed number. What would be the recommended minimum (stripped down) Freeswitch install to achieve this? We don't want to install elements that serve no functional purpose for this model nor do we want unneccessary processes consuming resources. Can someone give us some idea of sizing? Since the users will live in a database, the only sizing concern would be how many concurrent calls (assuming all are actively recording messages) for a given hardware configuration, say dual Xeon 2.8GHz with 4GB RAM? We did see the FAQ (How many concurrent calls can it support? Any benchmarks?), but were hoping for some real-world feedback, or even suggestions, if available, as we will be starting with several used servers of the aforementioned configuration. We had already built a relatively robust call recording application using Asterisk some time ago, and found that writing to ramdisk was the only way to achieve any worthwhile per-server performance, and as we are FreeSWITCH newbies, we decided it wouldn't hurt to ask. Cheers!

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FS and Asterisk connectivity
by Brian Snipes 24 Apr '08

24 Apr '08

I wish to connect FS to * for interoperability testing and can't seem to get my configs correct. Has anyone done this already and if so can you post your configs? I can call from FS to the asterisk side via a dialplan entry where ^9(.*)$ passes to a gateway I've setup: <extension name="out9_asterisk"> <condition field="destination_number" expression="^9(.*)$"> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> sip_profiles/default/asterisk.xml : <include> <gateway name="asterisk"> <param name="username" value="freeswitch"/> <param name="realm" value="x.x.x.x"/> <param name="password" value="password"/> <param name="register" value="false"/> </gateway> </include> The problem I get with this is no audio either way. I am on rev 8099. TIA, Brian

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Dialplan, default configuration. Dialing 1 goes to busy signal.
by Bryan Reynolds 24 Apr '08

24 Apr '08

I am new to Freewitch and VOIP in general so I apologize for the dumb questions in advance. Which of course makes it alright! I successfully downloaded and installed CentOS and compiled and installed Freeswitch. I have 3 polycom 501 phones connected and working. They can dial most of the features from the default dialplan, like conferencing, MOH, etc. When I dial 1 immediately goes to busy signal and has this output on the CLI. 2008-04-19 05:24:38 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1002(a)10.0.2.117 [45bf1fa0-66cc-4f1f-9e8d-6d45bd8ef391] 2008-04-19 05:24:38 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->1@default 2008-04-19 05:24:38 [NOTICE] switch_ivr.c:1032 switch_ivr_session_transfer() Transfer sofia/default/1002(a)10.0.2.117 to enum[1@default] 2008-04-19 05:24:38 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting 2008-04-19 05:24:38 [NOTICE] switch_core_state_machine.c:114 switch_core_standard_on_ring() Hangup sofia/default/1002(a)10.0.2.117 [CS_RING] [NO_ROUTE_DESTINATION] 2008-04-19 05:24:38 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 61 (sofia/default/1002(a)10.0.2.117) Ended 2008-04-19 05:24:38 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1002(a)10.0.2.117 [CS_HANGUP] This should not be the case because the regular expression below in the default plan should pick this up. I change the expression to "^9(10[01][0-9])$" and then I dialed 9 and the extension and it worked! Does anyone have an idea why that is happening? Is there some setting on the phone that makes it recognize 1 as something else? Here is the dialplan for local extensions. <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/>  <anti-action application="set" data="hangup_after_bridge=true"/>  <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/> <anti-action application="answer"/>  <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> Bryan Reynolds

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