April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Windows Build of latest git version
by Phillip Jones 22 Apr '10

22 Apr '10

Hi All, I just downloaded the lasted version from git, and I am having real problems compiling in VS2008 for Windows. FreeSwithCoreLib, mod_PortAudio and others are fine, but when building libsofia_sip_ua_static I get 905 errors, eg: Error 601 error C2065: 'sip_supported_hash' : undeclared identifier c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c 279 libsofia_sip_ua_static Error 602 error C2099: initializer is not a constant c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c 279 libsofia_sip_ua_static etc etc I have previously compiled successfully many times before, although not on this machine. Anyone else experience this and know how to fix? TIA Phil

2 1

Configure sofia-sip with --enable-ndebug
by Sergey Okhapkin 21 Apr '10

21 Apr '10

What is the best way to pass --enable-ndebug option (without editing FS files) to sofia-sip configure script? I'm getting FS crashes from time to time because of assertion failure in soa lib.

2 5

Responding to bridge after rejected attended transfer
by Troy Anderson 21 Apr '10

21 Apr '10

I have the latest version of FS (90913b8e26265fd381318334f40e0b1a038bb066 committed Apr 21) and am using the default config with a small change that allows me to respond to the various values originate_disposition values. First, is there a better way than what I am doing? Everything seems to work fine until I try it with an attended transfer. I've experienced this with Polycom, Cisco, and a soft phone client, so I don't think it's the phones, but who knows? With 3 extensions, say 1001, 1002, 1003, each on a different phone, I can call from 1001 to 1002. 1002 initiates a transfer to 1003. 1003 rejects so 1002 starts hearing voicemail. 1002 hits transfer again to "connect" 1001 to 1003's voicemail. With the default FS dialplan, it starts 1003's voicemail over, which is good. With this slight modification, it hangs up on 1001. If the feedback I get is that this modification looks fine, I can open a jira and supply a SIP trace and fs logs for the calls. Incidentally, to keep it simple, this example does the same thing regardless of the originate_disposition. In practice, there would be different actions taken. Thanks for any help! -Troy Here's what I have: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01]?[0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/>  <action application="transfer" data="BRIDGE-FOLLOWUP"/> </condition> </extension>  <extension name="BRIDGE-FOLLOWUP"> <condition field="destination_number" expression="BRIDGE-FOLLOWUP"/> <condition field="${originate_disposition}" expression="^SUCCESS$" break="on-true"> <action application="log" data="ERR (${originate_disposition})successful bridge is now done for ${dialed_extension}"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> <condition field="${originate_disposition}" expression="USER_BUSY|CALL_REJECTED" break="on-true"> <action application="log" data="ERR (${originate_disposition})busy/rejected for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition field="${originate_disposition}" expression="NO_ANSWER" break="on-true"> <action application="log" data="ERR (${originate_disposition})no answer for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition> <action application="log" data="ERR (${originate_disposition})other error for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> I also tried: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01]?[0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/>  </condition>  <condition field="${originate_disposition}" expression="^SUCCESS$" break="on-true"> <action application="log" data="ERR (${originate_disposition})successful bridge is now done for ${dialed_extension}"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> <condition field="${originate_disposition}" expression="^USER_BUSY|CALL_REJECTED$" break="on-true"> <action application="log" data="ERR busy/rejected for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition field="${originate_disposition}" expression="^NO_ANSWER$" break="on-true"> <action application="log" data="ERR no answer for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition> <action application="log" data="ERR other error for ${dialed_extension}"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension>

2 2

FreeSWITCH Conference Call
by Michael Collins 21 Apr '10

21 Apr '10

Hello all. Here is a reminder that the FreeSWITCH community conference call will be starting soon. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_21 Please feel free to add your agenda items. -MC

1 0

Voice quality Freeswitch on Centos on VMware EXSi
by Storm Demo 21 Apr '10

21 Apr '10

Dear All, I have installed Freesiwtch on CentOS 5.4 on VMware EXSi. Everything works fine but I keep having voice quality issues 1. Conference call will have gradually built up delay . Up to 10s of seconds for a conference call running for like 10 mins 2. Google talk have very choppy voice quality even I am using G711. I have extrated the RTPs in wireshark (less that 1 % packet lost) and play it using media player, the media is also choppy. For guest OS clock issue. I have already set clocksource=pit in linux boot option and NTP in linux but seems the problem still persist. Is there any suggestion that I can get the problem solved ? Or I have to run FS in real machine instead of visualization? Thanks a lot for your help

4 3

lua api_hangup_hook missing env?
by Troy Anderson 21 Apr '10

21 Apr '10

The wiki (http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) refers to the env global in mod_lua when executing a lua script set by api_hangup_hook, however, in practice, I don't see env set. Has this been changed or have I messed up somehow? I'm using the following from within the dial plan to get it going. <action application="set" data="api_hangup_hook=luarun recording.lua"/> <action application="bridge" data="user/107@${domain_name}" /> Here are the global variables lua does have set: 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: string 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: xpcall 2010-04-20 14:19:14.698522 [NOTICE] switch_cpp.cpp:1142 recording.lua: package 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: tostring 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: print 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: os 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: unpack 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: swig_type 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: require 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: getfenv 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: setmetatable 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: next 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: freeswitch 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: assert 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: argv 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: tonumber 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: io 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawequal 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: collectgarbage 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: getmetatable 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: module 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawset 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: ipairs 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: script_name 2010-04-20 14:19:14.699536 [NOTICE] switch_cpp.cpp:1142 recording.lua: swig_equals 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: math 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: debug 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: pcall 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: table 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: newproxy 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: type 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: coroutine 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _G 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: select 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: gcinfo 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: pairs 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: rawget 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: loadstring 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: tellme 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: _VERSION 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: dofile 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: setfenv 2010-04-20 14:19:14.700538 [NOTICE] switch_cpp.cpp:1142 recording.lua: load 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua: error 2010-04-20 14:19:14.701584 [NOTICE] switch_cpp.cpp:1142 recording.lua: loadfile Thanks! Troy

3 2

tiny bug in bootstrap
by jay binks 21 Apr '10

21 Apr '10

just downloaded latest from GIT.. ran bootstrap and got this : *./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf not found. You need autoconf version 2.59 or newer installed to build FreeSWITCH from SVN.* might need to change that so its saying SVN anymore.. -- Sincerely Jay

2 1

Re: [Freeswitch-users] Radius/Authentication/Authorization
by Abid Saleem 21 Apr '10

21 Apr '10

Dear Tihomir, It has been so many days since I did not get any response from your side. Could you please help me in this. Did you update the wiki for mod_rad_auth. If not, please help me in this regard. Thanks. Abid Saleem From: abid_freeswitch(a)live.com To: tculjaga(a)gmail.com; freeswitch-users(a)lists.freeswitch.org CC: neal(a)wanlink.com Subject: RE: [Freeswitch-users] Radius/Authentication/Authorization Date: Sat, 3 Apr 2010 12:02:55 +0500 Dear Tihomir, Any update and help on the below please. Abid From: abid_freeswitch(a)live.com To: tculjaga(a)gmail.com CC: neal(a)wanlink.com Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization Date: Wed, 31 Mar 2010 19:35:32 +0500 Dear Tihomir, Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps. Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows. 2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str** Please advise. Thanks. Regards-----------Abid Saleem --Forwarded Message Attachment-- From: tculjaga(a)gmail.com CC: neal(a)wanlink.com To: freeswitch-users(a)lists.freeswitch.org Date: Tue, 30 Mar 2010 21:42:03 +0200 Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization hello, here is an example <configuration name="rad_auth.conf" description="radius authentification module"> <settings>   </settings> <client> <param name="authserver" value="10.1.1.10:1812:gateway"/> <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/> <param name="seqfile" value="/var/run/radius.seq"/> <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/> <param name="default_realm" value=""/> <param name="radius_timeout" value="3"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="bindaddr" value="*"/> </client> <vsas>   <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/> <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/> <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/> <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/> <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/> <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>  <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/> <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/> <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/> <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/> <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/> <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/> <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>  </vsas> </configuration> in the dialplan you need to trigger auth as: <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> there are two behaviours: 1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number 2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/>  <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> <extension name="unitest_rad-ANI-balance"> <condition field="destination_number" expression="^602$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/> <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/> <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> will update the wiki by the end of the week. you have enough information for now. Tihomir. On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch(a)live.com> wrote: Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager Hotmail: Powerful Free email with security by Microsoft. Get it now. Hotmail: Powerful Free email with security by Microsoft. Get it now. _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969

1 0

Code negotiation - all codec offered
by Peter P GMX 21 Apr '10

21 Apr '10

Hello since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)" we see that - despite only allowing PMCA and PCMU codecs in vars.xml - Freeswitch offers all available codec to the called phone. Any idea how to solve this? Best regards Peter See Invite example U 10.xx.xx.1:5060 -> 10.xx.xx.172:2048 INVITE sip:31@10.xx.xx.172:2048;line=hxbudrul SIP/2.0. Via: SIP/2.0/UDP 10.xx.xx.1;rport;branch=z9hG4bKZD66c84339SHH. Max-Forwards: 70. From: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.1>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@10.xx.xx.1:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 920. X-FS-Support: update_display. Remote-Party-ID: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.1>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.1. s=FreeSWITCH. c=IN IP4 10.xx.xx.1. t=0 0. m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3 101 13. a=rtpmap:115 G7221/32000. a=fmtp:115 bitrate=48000. a=rtpmap:96 AMR/8000. a=fmtp:96 octet-align=0. a=rtpmap:99 SPEEX/32000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:7 LPC/8000. a=rtpmap:124 G726-16/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:6 DVI4/16000. a=rtpmap:123 G726-24/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:10 L16/22050. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode #

3 2

Two Major Problems
by Ken Fulmer 20 Apr '10

20 Apr '10

I'm using the external sip profile to route calls from a voice gateway router to an internal PBX. In separate xml files in the sofia/external directory, I have the following: <include> <gateway name="Server-1"> <param name="username" value="gateway"/> <param name="password" value="gateway"/> <param name="realm" value=" X.X.X.X "/> <param name="proxy" value=" X.X.X.X "/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> <include> <gateway name="Server-2"> <param name="username" value="gateway"/> <param name="password" value="gateway"/> <param name="realm" value="X.X.X.X"/> <param name="proxy" value="X.X.X.X"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> <include> <gateway name="Voice Gateway"> <param name="username" value="gateway"/> <param name="password" value="gateway"/> <param name="realm" value="X.X.X.X"/> <param name="proxy" value="X.X.X.X"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> In the dial plan/public directory, I have the following (different xml files): <extension name="PBX"> <condition field="destination_number" expression="^(Directory Numbers)$"> <action application="bridge" data="sofia/gateway/Server-1/$1"/> <action application="bridge" data="sofia/gateway/Server-2/$1"/> </condition> </extension> <extension name="PSTN"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="sofia/gateway/Voice Gateway/$1"/> </condition> </extension> In the sip profiles/eternal.xml file, I have the following major settings: <settings> <param name="inbound-proxy-media" value="false"/> <param name="disable-transcoding" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="user-agent-string" value="My Company Name"/> <param name="disable-register" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-codec-prefs" value="G729,PCMU" <param name="outbound-codec-prefs" value="G729,PCMU"/> <param name="media-option" value="resume-media-on-hold"/> <param name="stun-enabled" value="false"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5080"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> Currently, I'm experiencing two issues. 1. Inbound calls work as long as I'm using "inbound-proxy-media". When I disable that setting and force transcoding, the phone rings but immediately hangs up when I pick up the receiver. I see an "Incompatible Destination" message on the console. 2. Outbound calls don't work at all. On the console, I see "407 Proxy Authentication Required". If I'm using the external profile, how could it require authentication? Any ideas? Thanks, Ken Fulmer

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