- FreeSWITCH-users - FreeSWITCH Mailing Lists

Bypass_media
by Andre 30 Mar '14

30 Mar '14

I need a suggestion on how to bypass_media based on the gateway I'm using. Say we have 3 bridge's. One has to be full media but the other 2 can be bypass. If 1 fails, 2 will be the failover but since you have to set the media on leg a how do I change it to full media? Hangup after bridge Bridge 1 bypass_media = true Bridge 2 bypass_media = false Bridge 3 bypass_media = true

3 6

Using proxy with SIP trunk
by Jose Miguel Sucasas Mejuto 29 Mar '14

29 Mar '14

Hi all, I’m trying to set my SIP provider on my fresh installation of FreeSWITCH. My SIP provider needs to use a proxy and authenticated username, something like this: USERID: +3467******* DOMAIN: ims.vodafone.es AUTH. NAME: 214**************(a)ims.mnc001.mcc214.3gppnetwork.org PROXY: 62.87.93.90:5090 I have used this configuration in directory/default/example.com.xml: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}">  <param name="username" value="214**************(a)ims.mnc001.mcc214.3gppnetwork.org"/> <param name="password" value="******"/> <param name="from-user" value="+34672099215"/> <param name="from-domain" value="ims.vodafone.es"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="1001"/> <param name="contact-params" value="domain_name=$${domain}"/> <param name="context" value="public"/> <!—Added by me --> <param name="realm" value="ims.vodafone.es"/> <param name="proxy" value="62.87.93.90:5090"/>  </gateway> </gateways> <params> <param name="password" value="******"/> </params> </user> </include> but this configuration doesn’t work: The registration is OK: freeswitch@debrix> sofia status Name Type Data State ================================================================================================= 192.168.1.20 alias internal ALIASED internal profile sip:mod_sofia@192.168.1.20:5060 RUNNING (0) external profile sip:mod_sofia@192.168.1.20:5080 RUNNING (0) external::ims.vodafone.es gateway sip: 214**************@@ims.mnc001.mcc214.3gppnetwork.org@62.87.93.90:5090 REGED ================================================================================================= but when I try to make a call, I see the following: freeswitch@debrix> originate sofia/gateway/ims.vodafone.es/+34677****** &playback("hello world") 2014-03-29 20:06:40.634625 [NOTICE] switch_channel.c:1050 New Channel sofia/external/+34677****** [8b2f3c51-c95a-4b12-a82a-bcbb11f8626a] send 1046 bytes to udp/[62.87.93.90]:5090 at 19:06:40.981203: ------------------------------------------------------------------------ INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Contact: <sip:gw+ims.vodafone.es@47.62.8.53:5080;domain_name=192.168.1.20;transport=udp;gw=ims.vodafone.es> User-Agent: FreeSWITCH-mod_sofia/1.2.23-1~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 197 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@ims.vodafone.es>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1396103996 1396103997 IN IP4 47.62.8.53 s=FreeSWITCH c=IN IP4 47.62.8.53 t=0 0 m=audio 16004 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 289 bytes from udp/[62.87.93.90]:5090 at 19:06:40.985906: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" <sip:+34672099215@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE ------------------------------------------------------------------------ recv 508 bytes from udp/[62.87.93.90]:5090 at 19:07:13.017222: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090>;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Content-Length: 0 Contact: <sip:172.21.32.22:5070> P-Charging-Vector: icid-value=eb15701b0bdd73150b21ba59b77106f P-Charging-Function-Addresses: ccf="aaa://mm.ims.vodafone.es:3868;transport=tcp" ------------------------------------------------------------------------ send 334 bytes to udp/[62.87.93.90]:5090 at 19:07:13.017683: ------------------------------------------------------------------------ ACK sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090>;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 ACK Content-Length: 0 ------------------------------------------------------------------------ 2014-03-29 20:07:13.014627 [NOTICE] sofia.c:6659 Hangup sofia/external/+34677****** [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] -ERR ORIGINATOR_CANCEL 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1577 Session 5 (sofia/external/+34677******) Ended 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1581 Close Channel sofia/external/+34677****** [CS_DESTROY] I think that the error ( SIP/2.0 487 Request Terminated) is in the to-field of the header: INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> FreeSWITCH writes the proxy instead of the realm domain (ims.vodafone.es). Do you know how can I solve this problem? Thanks in advance, Regards

1 0

ZRTP enrollment lua script question
by Peter Villeneuve 29 Mar '14

29 Mar '14

Hi all, I'm trying out the option to have FS play trusted MiTM for ZRTP. I just compiled Master from Git and tried calling the 9787 extension from my softphone with ZRTP. I see in the dialplan that the script appears to get executed, however I don't hear anything at all and after a little while the call hangs up. It's not a codec issue because calling the IVR works fine. So, what else do I need to do to allow ZRTP enrollment? I see in the wiki that I need to set <action application="set" data="zrtp_enrollment=true"/> So how do I set this globally so that FS always negotiates ZRTP with enabled clients? Or do I need to set that up in the dialplan? 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000(a)my.domain.com Regex (PASS) [zrtp_enrollement] destination_number(9787) =~ /^9787$/ break=on-false 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000(a)my.domain.com Action lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/1000(a)my.domain.com) State Change CS_ROUTING -> CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/1000(a)my.domain.com[BREAK] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1000(a)my.domain.com) State ROUTING going to sleep 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000(a)my.domain.com) Running State Change CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1000(a)my.domain.com) State EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] mod_sofia.c:178 sofia/internal/1000(a)my.domain.com SOFIA EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:256 sofia/internal/1000(a)my.domain.com Standard EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000@my.domain.comhash(insert/my.domain.com-spymap/1000/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com hash(insert/my.domain.com-last_dial/1000/9787) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000@my.domain.comhash(insert/my.domain.com-last_dial/global/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com export(RFC2822_DATE=Fri, 28 Mar 2014 22:37:51 +0000) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_channel.c:1245 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 28 Mar 2014 22:37:51 +0000] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:343 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 d873ef12ce3143ba4ee41295c1b3199f49a89a78249ef454b3333a784d77acfd 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [opus:111:48000:20:0]/[opus:116:48000:20:0] Thanks, Peter

1 0

Freeswtich with freetdm - Span status is always "DOWN"
by Neo Cheema 29 Mar '14

29 Mar '14

I'm trying to use my ALLO Pri card with Libpri-Dahdi-FreeTDM stack. Here are the packages I've used Libpri - v1.4.14 Dahdi - v2.9.1 Freeswitch - Latest pull from Git All my efforts for Inbound/Outbound calls using PRI line are failing. freeswitch@caller0> originate FREETDM/1/1/998822xxxxx &park() 2014-03-29 14:59:44.346079 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:1 2014-03-29 14:59:44.346079 [NOTICE] switch_channel.c:1053 New Channel FreeTDM/1:1/9988225250 [e6b2e0fd-fa62-43eb-8b4b-0a48c79b0c99] 2014-03-29 14:59:44.366092 [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! 2014-03-29 14:59:44.366092 [NOTICE] mod_freetdm.c:1772 Close Channel FreeTDM/1:1/9988225250 [CS_INIT] 2014-03-29 14:59:44.366092 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [FREETDM] cause: [DESTINATION_OUT_OF_ORDER] -ERR DESTINATION_OUT_OF_ORDER Somehow the channels are not getting configured properly as the status of Span1 is always DOWN. Any help would be greatly appreciated. I've tried the following commands for loading/debugging etc. freeswitch@caller0> load mod_freetdm 2014-03-29 14:51:06.806089 [INFO] mod_enum.c:876 ENUM Reloaded 2014-03-29 14:51:06.806089 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-03-29 14:51:06.806089 [NOTICE] ftdm_sched.c:185 Launching main schedule thread 2014-03-29 14:51:06.806089 [NOTICE] ftdm_io.c:6344 Modules configured: 1 2014-03-29 14:51:06.806089 [NOTICE] ftmod_zt.c:1504 Using DAHDI control device 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5516 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:601 Setting rxgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:609 Setting txgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5480 Auto-loaded I/O module 'zt' 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:57 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 1:2 fd:58 .......... 2014-03-29 14:51:06.826081 [INFO] ftdm_io.c:5419 Configured 124 channel(s) 2014-03-29 14:51:06.826081 [CONSOLE] switch_loadable_module.c:1464 Successfully Loaded [mod_freetdm] 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'freetdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_ec' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'enable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm_usage' +OK Reloading XML +OK freeswitch@caller0> ftdm list <empty> freeswitch@caller0> ftdm start 1 +OK freeswitch@caller0> ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: B state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) Dahdi command outputs root@caller0:/usr/local/freeswitch/bin# dahdi_scan [1] active=yes alarms=OK description=ALLO (PCI) Quad E1 Card 0 Span 1 name=Tor3/0/1 manufacturer= devicetype= location=PCI Bus 02 Slot 01 basechan=1 totchans=31 irq=0 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 root@caller:/usr/local/freeswitch# lsdahdi ### Span 1: Tor3/0/1 "ALLO (PCI) Quad E1 Card 0 Span 1" (MASTER) CCS/HDB3/CRC4 ClockSource 1 unknown Clear 2 unknown Clear 3 unknown Clear 4 unknown Clear 5 unknown Clear 6 unknown Clear 7 unknown Clear 8 unknown Clear 9 unknown Clear 10 unknown Clear 11 unknown Clear 12 unknown Clear 13 unknown Clear 14 unknown Clear 15 unknown Clear 16 unknown HDLCFCS 17 unknown Clear 18 unknown Clear 19 unknown Clear 20 unknown Clear 21 unknown Clear 22 unknown Clear 23 unknown Clear 24 unknown Clear 25 unknown Clear 26 unknown Clear 27 unknown Clear 28 unknown Clear 29 unknown Clear 30 unknown Clear 31 unknown Clear

1 0

Playback case sensitive
by Grant Bagdasarian 28 Mar '14

28 Mar '14

Hello, Is there a way to make the playback application ignore case sensitivity of audio files it has to play? For instance: the actual file name is A.wav But when you do the following, it doesn't work because it can't find the file. <action application="playback" data="a.wav" /> It makes a sense that it doesn't. I'm just looking for a built-in freeswitch workaround/parameter/etc to get around this. Thanks, Grant

2 1

Mod Limit
by Andrew 28 Mar '14

28 Mar '14

Hi, I am trying to set a limit to one channel on leg A. For some reason this is not causing the call to fail and disconnect. Anyone have any ideas? leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}!NORMAL_TEMPORARY_FAILURE", 1,1)); leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}/1!NORMAL_TEMPORARY_FAILURE", 1, 1)); leg_A.SetVariable("limit_ignore_transfer", "true"); <transfer_history>1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfe r%3Alimit_exceeded/public/XML</transfer_history> <transfer_source>1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfer %3Alimit_exceeded/public/XML</transfer_source> <limit_usage>2</limit_usage> <limit_usage_outbound_gw_1>2</limit_usage_outbound_gw_1> <limit_rate>1</limit_rate> <limit_rate_outbound_gw_1>1</limit_rate_outbound_gw_1> <limit_ignore_transfer>true</limit_ignore_transfer>

1 0

Freeswitch/freetdm<->Freeswitch PCMU always wins
by Pete Ashdown 27 Mar '14

27 Mar '14

PSTN -> Freeswitch/freetdm (sending) -> Freeswitch/sip (receiving) PCMU is always winning between these two Freeswitch 1.2.23 boxes regardless of order in the codec list, when a call comes on freetdm. If I remove PCMU from the receiving box global_codec_prefs and thus the sofia external profile "CODECS IN" it works as expected and picks the top agreed codec from the freetdm sending box and the receiving. Before I paste debug logs, is there something I should be looking for on the sending box in relation to freetdm?

2 3

UBUNTU VS WINDOWS
by Ryan Gholam 27 Mar '14

27 Mar '14

Hi , I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! At Originate Result : *NORMAL_CLEARING, which is due to Socket Error .* *Is there some way i can have the same efficiency ? *

4 5

Shared Presence and jira ticket FS-3758
by Chris B. Ware 27 Mar '14

27 Mar '14

Hi guys, I've exactly the same problem stated on jira ticket FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance.

1 0

Shared Presence and jira ticket FS-3758
by Chris B. Ware 27 Mar '14

27 Mar '14

Hi guys, I've exactly the same problem stated on jira ticket FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance.

1 0