- FreeSWITCH-users - FreeSWITCH Mailing Lists

rtp timestamp issue on pickup
by Chris Martineau 22 Dec '11

22 Dec '11

Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris

2 4

Accept-Contact in sip INVITE
by Beppe Grillo 22 Dec '11

22 Dec '11

Hi, I need insert Accept-Contact in sip INVITE . I saw that RFC-3841 is supported . Do I configure something in the profile of gw to enter the Accept-contact in Sip Invite ? Regards, Giuseppe

1 0

One way voice on incoming calls
by Ryan V 22 Dec '11

22 Dec '11

Hi, We have configured a Sangoma A101DE. We are using Grandstream GXP280 phones and no NAT involved. We are using example configuration files except for one change in 00_inbound_did.xml. <action application="transfer" data="1004 XML default"/> Calls between extensions works fine and outbound calls also work fine. Though we are able to receive inbound calls, incoming audio is missing. Calling party hears us loud and clear. Any suggestions? Thanks, Ryan.

2 3

Re: [Freeswitch-users] Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error
by Peter Olsson 22 Dec '11

22 Dec '11

Any special reason you use the external profile for registrations? You will need to update the dialplan for the public context, so it knows how to route these calls, check the default dialplan for how It's handled there. /Peter ----- Reply message ----- Från: "Siobhan Hamilton" <siobhan(a)pluggedin-tech.com> Datum: tors, dec 22, 2011 07:44 Rubrik: [Freeswitch-users] Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error Till: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Hi there, I have a Freeswitch install I'm resurrecting from a while back that once worked, but seems to not be anymore. What I wanted to do, to start, is make sure that the machine was working properly. I fired it up and registered two clients on softphone apps, extensions 1002 and 1003; when I tried to get them to call one another, I got back a 404 error and a "No route, aborting"/NO_ROUTE_DESTINATION message in the logs. Does anyone have any clues? I am pretty much a complete newbie at troubleshooting these sorts of things. Thanks in advance for any help... My debug log is at the bottom, but first here is what I see when I check sofia status profile external; note that the clients appear and are registered (see below): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP xx.xxx.142.187 SIP-IP xx.xxx.142.187 URL sip:mod_sofia@xx.xxx.142.187:5060 BIND-URL sip:mod_sofia@xx.xxx.142.187:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN iLBC@30i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT iLBC@30i,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 19 FAILED-CALLS-IN 19 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: NDc0YWE1YTU1YjJhOTEwMGNmMGMxYjY2N2ZlYzMyMWY. User: 1003(a)xx.xxx.142.187 Contact: "1003" <sip:1003@192.168.1.108:5060;rinstance=da366cc7c432de35;transport=UDP> Agent: Zoiper rev.11619 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:50:06) EXPSECS(3038) Host: fswitch IP: 72.43.209.146 Port: 5060 Auth-User: 1003 Auth-Realm: xx.xxx.142.187 MWI-Account: 1003(a)xx.xxx.142.187 Call-ID: PfxBVFYxNIg.pWaY.CpEFO1EhEEqpMHJ User: 1002(a)xx.xxx.142.187 Contact: "user" <sip:1002@72.43.209.146:44803;transport=UDP;ob> Agent: CSipSimple r1108 / vivow-10 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:08:44) EXPSECS(556) Host: fswitch IP: 72.43.209.146 Port: 44803 Auth-User: 1002 Auth-Realm: xx.xxx.142.187 MWI-Account: 1002(a)xx.xxx.142.187 Total items returned: 2 ================================================================================================= Log (In debug mode) 2011-12-21 14:53:18.132608 [NOTICE] switch_channel.c:812 New Channel sofia/external/1003(a)xx.xxx.142.187 [96716f86-500e-445c-8196-59e754c023c0] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4659 Channel sofia/external/1003(a)xx.xxx.142.187 entering state [received][100] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4670 Remote SDP: v=0 o=Z 0 0 IN IP4 192.168.1.108 s=Z c=IN IP4 192.168.1.108 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:2757 Set Codec sofia/external/1003(a)xx.xxx.142.187 GSM/8000 20 ms 160 samples 13200 bits 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4616 Set 2833 dtmf send/recv payload to 101 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4837 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_NEW -> CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003(a)xx.xxx.142.187) State INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:84 sofia/external/1003(a)xx.xxx.142.187 SOFIA INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:124 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_INIT -> CS_ROUTING 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003(a)xx.xxx.142.187) State INIT going to sleep 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:1662 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change DOWN -> RINGING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003(a)xx.xxx.142.187) State ROUTING 2011-12-21 14:53:18.134757 [DEBUG] mod_sofia.c:147 sofia/external/1003(a)xx.xxx.142.187 SOFIA ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:77 sofia/external/1003(a)xx.xxx.142.187 Standard ROUTING 2011-12-21 14:53:18.134757 [INFO] mod_dialplan_xml.c:331 Processing 1003 <1003>->1002(a)xx.xxx.142.187 in context public Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->unloop] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [public_extensions] destination_number(1002(a)xx.xxx.142.187) =~ /^conf_(.*)$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->public_did] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [public_did] destination_number(1002(a)xx.xxx.142.187) =~ /^(5551212)$/ break=on-false 2011-12-21 14:53:18.134757 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2540 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change RINGING -> HANGUP 2011-12-21 14:53:18.134757 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/external/1003(a)xx.xxx.142.187 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2556 Send signal sofia/external/1003(a)xx.xxx.142.187 [KILL] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003(a)xx.xxx.142.187) State ROUTING going to sleep 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_HANGUP 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003(a)xx.xxx.142.187) State HANGUP 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:457 Channel sofia/external/1003(a)xx.xxx.142.187 hanging up, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 404 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:46 sofia/external/1003(a)xx.xxx.142.187 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003(a)xx.xxx.142.187) State HANGUP going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:351 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_HANGUP -> CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003(a)xx.xxx.142.187) State REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:53 sofia/external/1003(a)xx.xxx.142.187 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003(a)xx.xxx.142.187) State REPORTING going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:345 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_REPORTING -> CS_DESTROY 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1288 Session 18 (sofia/external/1003(a)xx.xxx.142.187) Locked, Waiting on external entities 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1306 Session 18 (sofia/external/1003(a)xx.xxx.142.187) Ended 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/1003(a)xx.xxx.142.187 [CS_DESTROY] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:449 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change HANGUP -> DOWN 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:452 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003(a)xx.xxx.142.187) State DESTROY 2011-12-21 14:53:18.138730 [DEBUG] mod_sofia.c:362 sofia/external/1003(a)xx.xxx.142.187 SOFIA DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:60 sofia/external/1003(a)xx.xxx.142.187 Standard DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003(a)xx.xxx.142.187) State DESTROY going to sleep !DSPAM:4ef2cf9132769289182587!

1 0

FreeSWITCH cluster with IM/Presence
by Even André Fiskvik 22 Dec '11

22 Dec '11

Hi. I currently have a project where I'm researching how to establish a clustered and scalable platform with: - registration - presence/blf between eyebeam/bria and Linksys SPA962 clients - IM between eyebeam/bria clients. - calling between the clients The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". How the registrations and calls etc. are balanced between them is not decided yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here? The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers (which can be scaled by adding n-nodes). I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, and what it would take to make something like this work. If there's any part of FS that is lacking some implementation to make such a scenario work, we are interested in providing the funding to support those features. I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would be helpful in this scenario or not. This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to. Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence. I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform. Best regards, Even André Fiskvik CTO - Oyatel AS

2 1

Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error
by Siobhan Hamilton 21 Dec '11

21 Dec '11

Hi there, I have a Freeswitch install I'm resurrecting from a while back that once worked, but seems to not be anymore. What I wanted to do, to start, is make sure that the machine was working properly. I fired it up and registered two clients on softphone apps, extensions 1002 and 1003; when I tried to get them to call one another, I got back a 404 error and a "No route, aborting"/NO_ROUTE_DESTINATION message in the logs. Does anyone have any clues? I am pretty much a complete newbie at troubleshooting these sorts of things. Thanks in advance for any help... My debug log is at the bottom, but first here is what I see when I check sofia status profile external; note that the clients appear and are registered (see below): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP xx.xxx.142.187 SIP-IP xx.xxx.142.187 URL sip:mod_sofia@xx.xxx.142.187:5060 BIND-URL sip:mod_sofia@xx.xxx.142.187:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN iLBC@30i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT iLBC@30i,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 19 FAILED-CALLS-IN 19 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: NDc0YWE1YTU1YjJhOTEwMGNmMGMxYjY2N2ZlYzMyMWY. User: 1003(a)xx.xxx.142.187 Contact: "1003" <sip:1003@192.168.1.108:5060;rinstance=da366cc7c432de35;transport=UDP> Agent: Zoiper rev.11619 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:50:06) EXPSECS(3038) Host: fswitch IP: 72.43.209.146 Port: 5060 Auth-User: 1003 Auth-Realm: xx.xxx.142.187 MWI-Account: 1003(a)xx.xxx.142.187 Call-ID: PfxBVFYxNIg.pWaY.CpEFO1EhEEqpMHJ User: 1002(a)xx.xxx.142.187 Contact: "user" <sip:1002@72.43.209.146:44803;transport=UDP;ob> Agent: CSipSimple r1108 / vivow-10 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:08:44) EXPSECS(556) Host: fswitch IP: 72.43.209.146 Port: 44803 Auth-User: 1002 Auth-Realm: xx.xxx.142.187 MWI-Account: 1002(a)xx.xxx.142.187 Total items returned: 2 ================================================================================================= Log (In debug mode) 2011-12-21 14:53:18.132608 [NOTICE] switch_channel.c:812 New Channel sofia/external/1003(a)xx.xxx.142.187 [96716f86-500e-445c-8196-59e754c023c0] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4659 Channel sofia/external/1003(a)xx.xxx.142.187 entering state [received][100] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4670 Remote SDP: v=0 o=Z 0 0 IN IP4 192.168.1.108 s=Z c=IN IP4 192.168.1.108 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:2757 Set Codec sofia/external/1003(a)xx.xxx.142.187 GSM/8000 20 ms 160 samples 13200 bits 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4616 Set 2833 dtmf send/recv payload to 101 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4837 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_NEW -> CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003(a)xx.xxx.142.187) State INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:84 sofia/external/1003(a)xx.xxx.142.187 SOFIA INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:124 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_INIT -> CS_ROUTING 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003(a)xx.xxx.142.187) State INIT going to sleep 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:1662 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change DOWN -> RINGING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003(a)xx.xxx.142.187) State ROUTING 2011-12-21 14:53:18.134757 [DEBUG] mod_sofia.c:147 sofia/external/1003(a)xx.xxx.142.187 SOFIA ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:77 sofia/external/1003(a)xx.xxx.142.187 Standard ROUTING 2011-12-21 14:53:18.134757 [INFO] mod_dialplan_xml.c:331 Processing 1003 <1003>->1002(a)xx.xxx.142.187 in context public Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->unloop] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [public_extensions] destination_number(1002(a)xx.xxx.142.187) =~ /^conf_(.*)$/ break=on-false Dialplan: sofia/external/1003(a)xx.xxx.142.187 parsing [public->public_did] continue=false Dialplan: sofia/external/1003(a)xx.xxx.142.187 Regex (FAIL) [public_did] destination_number(1002(a)xx.xxx.142.187) =~ /^(5551212)$/ break=on-false 2011-12-21 14:53:18.134757 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2540 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change RINGING -> HANGUP 2011-12-21 14:53:18.134757 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/external/1003(a)xx.xxx.142.187 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2556 Send signal sofia/external/1003(a)xx.xxx.142.187 [KILL] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003(a)xx.xxx.142.187) State ROUTING going to sleep 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_HANGUP 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003(a)xx.xxx.142.187) State HANGUP 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:457 Channel sofia/external/1003(a)xx.xxx.142.187 hanging up, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 404 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:46 sofia/external/1003(a)xx.xxx.142.187 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003(a)xx.xxx.142.187) State HANGUP going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:351 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_HANGUP -> CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003(a)xx.xxx.142.187) State REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:53 sofia/external/1003(a)xx.xxx.142.187 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003(a)xx.xxx.142.187) State REPORTING going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:345 (sofia/external/1003(a)xx.xxx.142.187) State Change CS_REPORTING -> CS_DESTROY 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003(a)xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1288 Session 18 (sofia/external/1003(a)xx.xxx.142.187) Locked, Waiting on external entities 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1306 Session 18 (sofia/external/1003(a)xx.xxx.142.187) Ended 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/1003(a)xx.xxx.142.187 [CS_DESTROY] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:449 (sofia/external/1003(a)xx.xxx.142.187) Callstate Change HANGUP -> DOWN 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:452 (sofia/external/1003(a)xx.xxx.142.187) Running State Change CS_DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003(a)xx.xxx.142.187) State DESTROY 2011-12-21 14:53:18.138730 [DEBUG] mod_sofia.c:362 sofia/external/1003(a)xx.xxx.142.187 SOFIA DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:60 sofia/external/1003(a)xx.xxx.142.187 Standard DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003(a)xx.xxx.142.187) State DESTROY going to sleep

1 0

Forwarding MWI from gateway to internal extensions
by Gabriel Burca 21 Dec '11

21 Dec '11

I just started using FreeSWITCH yesterday. Everything works perfectly, except I can't figure how to forward a SIP message waiting indication from a gateway to an internal extension. This is a simple home use-case. I have FS connected externally to a VoIP provider. Incoming calls (from the provider) are routed to a group extension which rings all phones. VM is handled by the VoIP provider. I see the SIP NOTIFY message come in to FS, but I don't know how to pass that notification message on to extension 1000 for example, or better yet, to all the extensions that are part of a group. I've added: <param name="forward-unsolicited-mwi-notify" value="true"/> to my external gateway config file, but it doesn't do the trick, and even if it did, I don't see how it would know where to forward the MWI to. Don't know if this is related, but I also see the following in the logs: [DEBUG] sofia.c:445 Gateway information missing Subscription Event: message-summary I've gone through most of the relevant documentation, and searched on-line, but came up empty handed. Is this something that can be accomplished through configuration files (and if so, how?) or does this require scripting? -- Gabriel Burca There's nothing remarkable about it. All one has to do is hit the right keys at the right time and the instrument plays itself. -- J. S. Bach

1 0

Issue adding a SIP Gateway
by Ryan Watkins 21 Dec '11

21 Dec '11

Hello all, Having an issue adding a gateway, and hoping someone can help. Please forgive me... still new to Freeswitch and XML, but I've exhausted all that I know, so I need to reach out. Running Ubuntu, and I installed Freeswitch following the wiki guide. Didn't appear to have any issues installing, didn't get any noticeable errors during the setup anyways. Been following the Freeswitch 1.0.6 book to add a gateway (using iptel) I've created the iptel.org.xml in both /opt/freeswitch/conf/sip_profiles/external and /usr/src/freeswitch/conf/sip_profiles/external However I am unable to get the gateway to load in fs_cli Looking at all the log data, it only gives notice regarding the example.com gateway... does not appear to mention iptel at all, and no err log shows regarding iptel I attempted to manually unregister/register iptel, but I get the following error when I try sofia profile external unregister iptel (same error with register iptel) Invalid gateway! Any ideas? Thanks Ryan Watkins Ryan Watkins Networking & Customer Support OCENS 19655 1st AVE S. #203 Seattle, WA 98148 Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH ________________________________________________________ www.ocens.com<http://www.ocens.com/> | support.ocens.com<http://support.ocens.com/> Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314

4 11

Sofia late-negotiation on re-INVITE (codec-modification)
by Visciano Giovanni 21 Dec '11

21 Dec '11

Basic call SIP vs SIP with: <param name="inbound-late-negotiation" value="true"/> <action application="set" data="inherit_codec=true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> This help to avoid FS transcoding, the SIP endpoints negotiate codec end-to-end. FS still in the media path but no transcoding applied. INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] If a SIP ep send a re-INVITE for codec modification, sofia negotiate the new codec on that leg without "asking" the remote party. This result in transcoding and since I have mod_g729 only in passthrough mode the call is released by the core. re-INVITE [g729] ---> FS 200 OK [g729] <--- FS ************* It seems that "late_negotiation + inherit_codec" doesn't apply to a codec modification. Do I miss some configuration options in the SIP profile? (Note: I cannot use proxy-media/no-media mode in my configuration) ************* Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

4 8

FreeSWITCH Conference Call Today
by Michael Collins 21 Dec '11

21 Dec '11

Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_21 We will be talking about CDRs today. DRK will be sharing some of his experience with parsing and rating. If anyone else has experience with CDR parsing and rating please give us your input. We'll also be discussing some CDR collection techniques. Lastly, we have a few updates for the community, including Mitch Capper discussing some TLS changes in FreeSWITCH. Talk to you soon! -Michael

1 0