December 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Farsi language support in FreeSwitch
by Chandramouli P 02 Jan '17

02 Jan '17

Hello, I am planning to use FreeSwitch to implement a small project. I have the below queries: 1) Does FreeSwitch supports "Farsi" language? 2) Is it possible to say current date and time in "Farsi" language" 3) Is it possible to calculate the active call length in FreeSwitch? I mean that let us assume ten calls are active on a single channel (concurrent calls) and they are listening my welcome greeting or file. Now, I want to calculate the active call length of the users. Is it possible or not? Thanks in advance, Chandramouli.

3 5

Memory leak
by Sergey Safarov 30 Dec '16

30 Dec '16

I use monitoring tool (zabbix) to check health of FreeSwitch servers. I has found that after 60 days of uptime FS host has less free memory than after server reboot. This memory is returned back after FS daemon restart. On FS host is installed 66fe6e5. How to correctly detect memory leak in FS?

6 19

Re: [Freeswitch-users] [ERR] switch_ivr_originate.c:224 sofia/sipinterface_1/ Error Playing File!
by Michael Dunton 29 Dec '16

29 Dec '16

Hi Vallimamod, Thank you for your reply. Here is the information you requested. I've masked some of the data. freeswitch-debug 2016-12-29 18:12:51.367469 [NOTICE] kazoo_node.c:302 log|691c7b7e-cdf2-11e6-9852-a73efaac9f3b|building xferext extension: bridge {ignore_display_updates='true',ecallmgr_Account-ID='98c46437cbcd43e3d1a518a8de083038',ecallmgr_Reseller-ID='0534605f933714aca28e2b34b49e2a73',effective_caller_id_number='+1MASKED_NUMBER2',effective_caller_id_name='1MASKED_NUMBER2',origination_caller_id_number='+1MASKED_NUMBER2',origination_caller_id_name='1MASKED_NUMBER2',ignore_early_media='true',local_var_clobber='true'}[sip_from_uri='sip:+1MASKED_NUMBER2@MA.SK.ED.IP2',ecallmgr_Global-Resource='true',ecallmgr_Resource-ID='bandwidth',ecallmgr_Original-Number='MASKED_NUMBER',leg_progress_timeout='8',effective_callee_id_number='MASKED_NUMBER',effective_callee_id_name='MASKED_NUMBER']sofia/sipinterface_1/+1MASKED_NUMBER(a)MA.SK.ED.IP2|[sip_from_uri='sip:+1MASKED_NUMBER2@MA.SK.ED.IP',ecallmgr_Global-Resource='true',ecallmgr_Resource-ID='bandwidth',ecallmgr_Original-Number='MASKED_NUMBER',leg_progress_timeout='8',effective_callee_id_number='MASKED_NUMBER',effective_callee_id_name='MASKED_NUMBER']sofia/sipinterface_1/+1MASKED_NUMBER(a)MA.SK.ED.IP freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:12:51.367469 [NOTICE] switch_channel.c:1077 New Channel sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2 [694ca916-cdf2-11e6-9863-a73efaac9f3b] freeswitch-debug 2016-12-29 18:12:51.390874 [INFO] kazoo_node.c:625 exec: uuid_setvar_multi(694ca916-cdf2-11e6-9863-a73efaac9f3b ecallmgr_Channel-Authorized=true;ecallmgr_Global-Resource=true;ecallmgr_Reseller-ID=0534605f933714aca28e2b34b49e2a73;ecallmgr_Account-ID=98c46437cbcd43e3d1a518a8de083038) freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:12:52.147490 [NOTICE] sofia.c:6767 Ring-Ready sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2! freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:13:19.227463 [NOTICE] sofia.c:7532 Channel [sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2] has been answered freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:13:22.007462 [NOTICE] switch_ivr_originate.c:3519 Hangup sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2 [CS_CONSUME_MEDIA] [NO_ANSWER] freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:13:22.007462 [ERR] switch_ivr_originate.c:224 sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2 Error Playing File! freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:13:22.007462 [NOTICE] switch_core_session.c:1642 Session 14305 (sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2) Ended freeswitch-debug 694ca916-cdf2-11e6-9863-a73efaac9f3b 2016-12-29 18:13:22.007462 [NOTICE] switch_core_session.c:1646 Close Channel sofia/sipinterface_1/%2B1MASKED_NUMBER(a)MA.SK.ED.IP2 [CS_DESTROY] Thanks, Michael Dunton Software Developer Tresta.com >Hi John, > >Can you please give more details on how this error appears? Your dialplan and debug logs will be helpful. > >In the mean time, I have noticed a few things that may help you: > >- The current stable version of freeSWITCH is 1.6, You should definitely try with it. The source and repos are available at https://freeswitch.org/>confluence/display/FREESWITCH/Installation > >- Unless you have altered it, the channel name "sofia/sipinterface_1/" is incomplete, it should be like "sofia/sipinterface_1/dialed_extension". You >may need to double check that your dial params are correct and the call was actually made. > >- The timeout variables are for playback terminators (when the user presses a key to stop the playback). So they won't help if the call never made it. > >- When the call is hanged up during the playback, it is not considered as an error by switch_ivr_play_file() > > >Best Regards, >-- >Vallimamod Abdullah >vma at vallimamod.org >. > >> On 20 Dec 2016, at 22:41, John.Nowlin at tresta.com wrote: >> >> FreeSWITCH Version 1.4.26~64bit ( 64bit) >> >> We receive this error, “[ERR] switch_ivr_originate.c:224 sofia/sipinterface_1/ Error Playing File!” whenever the leg receiving the call hangs up before the extant media can start playing. >> >> After looking through switch_ivr_originate.c, I tried setting the global variables playback_timeout_sec=10 and playback_timeout_as_success=true but did not see a change in behavior. >> >> status = switch_ivr_read(collect->>session, >> (uint32_t)len, >> (uint32_t)len, >> collect->>file, NULL, buf, sizeof(buf), collect->>confirm_timeout, NULL, 0); >> >> >> if (status != SWITCH_STATUS_SUCCESS && status != SWITCH_STATUS_BREAK && status != SWITCH_STATUS_TOO_SMALL) { >> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(collect->>session), SWITCH_LOG_ERROR, "%s Error Playing File!\n", >> switch_channel_get_name(channel)); >> switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); >> } >> >> Is there some other setting that can inform switch_ivr_originate to not worry about having not played the media after the recipient has hung up? >> >> john nowlin >> Senior Software Developer >> >> Tresta >> office: 850.383.7756 >> mobile: 850.294.5635 >> www.tresta.com >> — >> A Better Phone Experience >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users MA.SK.ED.IP2 http://www.freeswitch.org

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FS changes to transcoding after hold
by Sebastian Kemper 29 Dec '16

29 Dec '16

Hello all, I have this Gigaset C430 SIP phone connected to FreeSWITCH 1.6.13. The phone it configured to offer G722 and G711. I use this on both internal and external SIP profile: <param name="inbound-codec-negotiation" value="greedy"/> <param name="inbound-late-negotiation" value="true"/> On the internal profile I also have the following, because the phone plays its own MOH so FS doesn't have to: <param name="disable-hold" value="true"/> So now I make an outbound call from the phone. It offers G722 and G711. FS offers both on the egress leg. On the egress G711 is negotiated. FS negotiates G711 with the Gigaset afterward. All fine. Next I push the R button on the phone and it sends a reINVITE to FS: INVITE sip:500@172.31.255.129:5892;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.31.255.132:5060;branch=z9hG4bK3acb589137e2c87caa513ddc3c92fb1d;rport From: <sip:301@172.31.255.129>;tag=3900142461 To: <sip:500@172.31.255.129;user=phone>;tag=6SX155B96H1pp Call-ID: 1793314766@172_31_255_132 CSeq: 4 INVITE Contact: <sip:301@172.31.255.132:5060> Proxy-Authorization: Digest username="301", realm="172.31.255.129", qop=auth, algorithm=MD5, uri="sip:500@172.31.255.129:5892;transport=udp", nonce="e68c4915-ef56-4f0a-ad28-f8c8535731c1", nc=00000002, cnonce="64b2c2e298dce6f8a5a79c06598c42f0", response="d9ab62f9b94158000dbf92dfe31a6a10" Max-Forwards: 70 User-Agent: C430A GO/42.240.00.000.000 Supported: replaces Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE Content-Type: application/sdp Content-Length: 143 v=0 o=301 5012 33 IN IP4 172.31.255.132 s=Mapping c=IN IP4 172.31.255.132 t=0 0 m=audio 5012 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendonly OK, so here it says it's going to send MOH and tells FS to listen. Next FS replies with 200-OK. FS does not send anything on the egress leg of this bridge (fine with me). SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.255.132:5060;branch=z9hG4bK3acb589137e2c87caa513ddc3c92fb1d;rport=5060 From: <sip:301@172.31.255.129>;tag=3900142461 To: <sip:500@172.31.255.129;user=phone>;tag=6SX155B96H1pp Call-ID: 1793314766@172_31_255_132 CSeq: 4 INVITE Contact: <sip:500@172.31.255.129:5892;transport=udp> Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: timer, path, replaces Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 195 v=0 o=root 1481307576 1481307578 IN IP4 172.31.255.129 s=root c=IN IP4 172.31.255.129 t=0 0 m=audio 10010 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=recvonly After a while I cancel the hold on the phone. Remember, G711 was negotiated on the initial bridge. INVITE sip:500@172.31.255.129:5892;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.31.255.132:5060;branch=z9hG4bK59938e08302f7ed7f1b3f1005aab27c3;rport From: <sip:301@172.31.255.129>;tag=3900142461 To: <sip:500@172.31.255.129;user=phone>;tag=6SX155B96H1pp Call-ID: 1793314766@172_31_255_132 CSeq: 5 INVITE Contact: <sip:301@172.31.255.132:5060> Proxy-Authorization: Digest username="301", realm="172.31.255.129", qop=auth, algorithm=MD5, uri="sip:500@172.31.255.129:5892;transport=udp", nonce="e68c4915-ef56-4f0a-ad28-f8c8535731c1", nc=00000003, cnonce="9bb91b66139de548b11790f4ccf2a838", response="12c5a33918a6890c0a675177c556239b" Max-Forwards: 70 User-Agent: C430A GO/42.240.00.000.000 Supported: replaces Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE Content-Type: application/sdp Content-Length: 223 v=0 o=301 5012 34 IN IP4 172.31.255.132 s=Mapping c=IN IP4 172.31.255.132 t=0 0 m=audio 5012 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 So the phone offers again G722 and G711. And now FS replies with this: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.255.132:5060;branch=z9hG4bK59938e08302f7ed7f1b3f1005aab27c3;rport=5060 From: <sip:301@172.31.255.129>;tag=3900142461 To: <sip:500@172.31.255.129;user=phone>;tag=6SX155B96H1pp Call-ID: 1793314766@172_31_255_132 CSeq: 5 INVITE Contact: <sip:500@172.31.255.129:5892;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: timer, path, replaces Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 195 v=0 o=root 1481307576 1481307579 IN IP4 172.31.255.129 s=root c=IN IP4 172.31.255.129 t=0 0 m=audio 10010 RTP/AVP 9 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Now the call is going on with transcoding from G711 to G722. I see it both in the increase in CPU usage (FS is running on a tiny for-home-use MIPS router) and with "show detailed_calls" where it showed read_codec, write_codec, b_read_codec and b_write_codec as PCMA before Hold, and read_codec and write_codec G722 and b_read_codec and b_write_codec PCMA after Hold. It's probably not the smartest move for the phone to renegotiate codecs at this point. But I think FS should be able to deal with it (without simply falling back to transcoding). I've tried if putting "disable-transcoding" into the profiles changes anything, but it didn't. "renegotiate-codec-on-hold" and "renegotiate-codec-on-reinvite" are gone since FS 1.6.10 so I can't try those. I read something about "indicate_hold" which probably means that FS will send a reINVITE on the other leg, so I might try that. Does anyone have any other thoughts on this? Kind regards, Sebastian

4 5

Configure Gateway Auth username
by Murugan Pandian 29 Dec '16

29 Dec '16

Hi, I am using following sip trunk detail in my xlite it work fine and i can able to make call but when i use with my freeswitch in gateway ,I am always getting 403 error i understand have problem in my gateway config xml. SIP TRUNK details (Examples): Username: trunkusername(a)bangalore.relianceims.in Password: trunkpassword outbound proxy: 10.21.254.231 Bellow is my reliancegateway.xml <include> <gateway name="reliance"> <param name="realm" value="bangalore.relianceims.in" /> <param name="username" value="trunkusername" /> <param name="password" value="trunkpassword" /> <param name="proxy" value="10.21.254.231" /> <param name="register" value="true" /> <param name="ping" value="5" /> <param name="ping-max" value="3" /> <param name="retry-seconds" value="5" /> <param name="expire-seconds" value="60" /> <param name="context" value="public"/> <variables> <variable name="verbose_sdp" value="true"/> </variables> </gateway> </include> Is there any way to use Auth username in gateway config xml

3 4

Remove a user from a call group if error
by Antonio Silva 29 Dec '16

29 Dec '16

Hi, is it possible to remove a user from a call group if the dialstring or hangup cause returns error similar to the option "outbound_redirect_fatal". I've a group "200" with members "1" and "2", I then call to the group using call_group(200+F) if the user 1 is not registered just call the user 2. Now ii see calling user 1 that is immediately hangup with hangup_cause user_not_registered. Write now i've lua script that executes the api call_group(200+F), parses the dialtring and remove the error/not_registered before executing the command bridge(dialstring), i wanted to avoid using a lua script to call a group. Can it be done? -- Saludos / Regards / Cumprimentos, António silva

1 0

(no subject)
by Hardik Patel 28 Dec '16

28 Dec '16

I am trying to configure G729 codec in freeswitch but getting this error while i am trying to install it. Initially i have configure it successfully but then its required to upgrade Freeswitch and after up-gradation it am trying to reconfigure it and getting error. ERROR [this code has been activated too many times, see http://wiki.freeswitch.org/wiki/Mod_com_g729#Saving_Config_Files] Any one can help me please? -- Hardik Patel iNextrix Technologies Pvt Ltd

3 2

Voicemail MWI(message waiting indicator) issue
by Ketan Kothari 27 Dec '16

27 Dec '16

Hello, [image: Inline image 1] Devices look like this in, sofia status profile internalprofile reg Phone- B Call-ID: 44461523-5062-2(a)BJC.BGI.B.BGB User: 1519124(a)172.30.0.20 Contact: "" <sip:1519124@12.72.14.16:5062> Agent: Grandstream GXP1620 1.0.0.38 Status: Registered(UDP)(unknown) EXP(2016-12-27 13:21:50) EXPSECS(3344) Ping-Status: Reachable Ping-Time: 0.00 Host: ip-172-30-0-20 IP: 12.72.14.16 Port: 5062 Auth-User: 911519124 Auth-Realm: 54.14.15.64 MWI-Account: 1519124(a)172.30.0.20 Phone- A Call-ID: MTFmYWQyN2RmZDNmZTZiNzgzM2E5ZjVkOGIzMDI4NTQ. User: 911519124(a)172.30.0.20 Contact: "" <sip:911519124@12.72.14.16:51018 ;rinstance=ffa5b67a6c872492;transport=UDP> Agent: Z 3.3.25608 r25552 Status: Registered(UDP)(unknown) EXP(2016-12-27 13:27:00) EXPSECS(3654) Ping-Status: Reachable Ping-Time: 0.00 Host: ip-172-30-0-20 IP: 12.72.14.16 Port: 51018 Auth-User: 911519124 Auth-Realm: 54.14.15.64 MWI-Account: 911519124(a)172.30.0.20 Devices look like this in, show registrations Phone-B 1519124,172.30.0.20,44461523-5062-2(a)BJC.BGI.B.BGB,sofia/internalprofile/ sip:1519124@12.72.14.16:5062,1482844910,12.72.14.16,5062,udp,ip-172-30-0-20, Phone- A 911519124,172.30.0.20,MTFmYWQyN2RmZDNmZTZiNzgzM2E5ZjVkOGIzMDI4NTQ.,sofia/internalprofile/sip:911519124@12.72.14.16:51018 ;rinstance=ffa5b67a6c872492;transport=UDP,1482845220,12.72.14.16,51018,udp,ip-172-30-0-20, When device C drops voicemail to 911519124. The phone A (with sip user : 911519124) notify with MWI but phone B (with sip user : 1519124) not blinking (no MWI). Please let me know what am i missing here?

2 1

different ringtone when called party is already busy on another call
by Bipin Patel 26 Dec '16

26 Dec '16

3 13

How to edit voicemail IVR. How to remove IVR Menu option "To mark this message urgent press *"
by Naveen Tamanam 26 Dec '16

26 Dec '16

Hi, Freeswich dialplan appliction voicemail plays the IVR and it provides the several options after we leave the voicemail. How to remove the menu options like "To mark this message urgent press *" after we leave the voicemail. -- Thanks & Regards, Naveen Tamanam

1 0