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Polycom horror story
by Luis Daniel Lucio Quiroz 31 Dec '14

31 Dec '14

Hello, Using Freeswitch 1.4.14 with multitenant, I can not make this @#$@%# telephone to register: model: Soundpoint IP501 firmware: 3.1.8.0070 I have done next: NDLB-force-rport with true and safe values No nat involved (switch and phone on same segment) I have tried multiple combinations of configurations on the phone (using web interface). with or without proxy, different sip extensions, different passwords, different ports. I always get 403 error code. however, i can register only if i do next: after telephone is booted, i edit sip password using the phone keypad. And telephone register. This makes me thing it is a bug on the polycom and has nothing to do with Freeswitch. It works on all other phones I have, snom, bria, linphone. But here I am, maybe someone has workarround this somehow. LD

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Flash Hook to an FXO port (ata)
by davy van de moere 31 Dec '14

31 Dec '14

Gents, Does any one have a good setup to send a flash hook to an analogue line? Preferably with an ATA of some sort. So far I've been trying with an Linksys SPA3102, which can do a flash hook with the specific mime type : application/hook-flash , with Freeswitch I the send_dtmf didn't get me any further than sending the special flash key F in application/dtmf-relay. Which is not interpreted by the SPA3102 as a flash hook. Hence, does anyone have a good strategy from a Freeswitch pov to send a flash hook signal (do I need to think sipsak?), or an ATA/firmware which does support the F key in dtmf? Season grtz! Davy

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intercom setup -- page multiple phones in a single endpoint
by Vik Killa 31 Dec '14

31 Dec '14

im using "conference_set_auto_outcall" to do an intercom if i set "conference_set_auto_outcall" to "user/1000" and that endpoint points to two physical phones, only one is picked up, the other originate is canceled by FS Example: user/1000 ==> 1000(a)192.168.0.101 and 1000(a)192.168.0.102 does anyone know how to address this so that both phones are placed in the conference? is there a flag to pass to the originate to tell it NOT to cancel the second call? Thanks. /V

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Trouble Playing Audio to external dialler after bridged call to external SIP URI is hung up
by Notify Me 31 Dec '14

31 Dec '14

Hi! I am still a Freeswitch Newbie, and I am trying to learn by doing. I have installed and using version 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit on a 64-bit CentOS 6.6 kvm node. I have been able to successfully transfer calls for an external dialler, through a SIP trunk, to call an external SIP URI. This works and I can see it in the logs as very successful. The external SIP URI is supposed to process the caller_id, and send the dialler an SMS. The problem is the SIP URI being dialled ends the call very quickly and the external dialler has a busy tone and the call is dropped, or other signals that do not indicate that the call was successful and may dial again several times, if the SMS does not get delivered in time. The SIP URI sends back a 100 Trying and a 480 Temporarily Unavailable response as you can see below from a dump of the sequence. I have tried to introduce ivr sounds before and after the user connects and the bridge is dropped but is also not working. I can see that the wav files are being called from the logs but I dont hear anything at all. Please can anyone guide me? my dialplan: public: <include> <extension name="inbound"> <condition field="destination_number" expression="^012345(\d{2})$"/> <action application="set" data="ringback=${us-ring}"/> <condition field="caller_id_number" expression="^0(\d{10})$" require-nested="true"> <action application="set" data="effective_caller_id_number=+234$1"/> <action application="transfer" data="1212 XML default"/> </condition> </extension> </include> default: <extension name="1212"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="transfer_after_bridge=1213:XML:default"/> <action application="set" data="exec_after_bridge_app=transfer"/> <action application="set" data="exec_after_bridge_arg="1213"/> <action application="bridge" data="sofia/external/sip:user.name@sip.othersipgw.ip.address.com;${effective_caller_id_number}"/> </condition> </extension>  <extension name="1213"> <condition field="destination_number" expression="1213"> <action application="set" data="instant_ringback=true"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="playback" data="ivr/ivr-thank_you.wav"/> <action application="playback" data="ivr/ivr-thank_you.wav"/> </condition> </extension> TCP DUMP 09:31:58.808760 IP (tos 0x0, ttl 64, id 34090, offset 0, flags [none], proto UDP (17), length 1146) my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:user.name@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 5 From: "0diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=6pKBvpXc32yeD To: <sip:user.name@sip.othersipgw.com;+234diallednumber> Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE Contact: <sip:mod_sofia@my.public.ip.address:5080> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "0diallednumber" <sip:+234diallednumber@my.public.ip.address>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418006836 1418006837 IN IP4 my.public.ip.address s=FreeSWITCH c=IN IP4 my.public.ip.address t=0 0 m=audio 20682 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 09:31:59.080097 IP (tos 0x0, ttl 53, id 5734, offset 0, flags [none], proto UDP (17), length 384) othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, length: 356 SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr From: "0diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=6pKBvpXc32yeD To: <sip:user.name@sip.othersipgw.com;+234diallednumber> Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Content-Length: 0 09:31:59.145629 IP (tos 0x0, ttl 53, id 5735, offset 0, flags [none], proto UDP (17), length 1015) othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, length: 987 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 4 From: "0diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=6pKBvpXc32yeD To: <sip:user.name@sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: user.name X-FS-Display-Number: sip:user.name@sip.othersipgw.com Remote-Party-ID: "user.name" <sip:user.name@sip.othersipgw.com>;party=calling;privacy=off;screen=no 09:31:59.146980 IP (tos 0x0, ttl 64, id 34091, offset 0, flags [none], proto UDP (17), length 412) my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:user.name@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 5 From: "0diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=6pKBvpXc32yeD To: <sip:user.name@sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 ACK Content-Length: 0

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Re: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI
by Sina Owolabi 31 Dec '14

31 Dec '14

Avi Thanks for all your assistance. I found out why I wasnt hearing anything. For some reason I had to put the full path to the wav file before I would hear anything played. I am also not transferring to another extension now, thanks to more help I got in irc. Everything happens in the public inbound extension like so: <extension name="transfer2external"> <condition field="${caller_id_number}" expression="^01234(\d{6})$" require-nested="false"> <action application="set" data="effective_caller_id_number=01234${1}"/> <action application="set" data="effective_caller_id_name=01234${1}"/> </condition> <condition field="destination_number" expression="^012345(?!85|90|91|92)(\d)(\d)$"> <action application="ring_ready" /> <action application="sleep" data="5000"/> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> <action application="bridge" data="sofia/external/sip:username@sip.othersipgw.com;${effective_caller_id_number}"/> </condition> </extension> I get a ring and the call is transferred. Thanks again! On Wed, Dec 10, 2014 at 12:13 PM, Avi Marcus <avi(a)avimarcus.net> wrote: > Your log still shows: Action set(exec_after_bridge_arg=1213) > > ... and if they set a 180, then you can set the ringback. > -Avi

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FreeSWITCH Week in Review (Master Branch) December 21st-27th
by Ken Rice 30 Dec '14

30 Dec '14

New Post on freeswitch.org from kathleen check it out at http://freeswitch.org/freeswitch-week-in-review-master-branch-december-21st… FreeSWITCH Week in Review (Master Branch) December 21st-27th Hello, again. This week in the FreeSWITCH master branch we had 20 commits. The feature for this week is allowing streaming binary data from mod_memcache. The commit author, Travis Cross, describes this improvement as “Previously data returned from `memcache get` would be truncated at the first NULL byte. By using raw_write_function here to stream the returned memcache value, we allow mod_memcache to be used for audio and other arbitrary binary data.” New features that were added: 750b1dd FS-7114 Allow streaming binary data from mod_memcache These were the packaging improvements: 3c8dd3e Handle missing `lsb_release` 505cd29 Refactor distro detection and handling 430433a Improve error message d88bae1 Support optional debian parallel builds Improvements in cross platform build supports: af6b23a FS-7046 Fix some additional Windows build warnings for mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] The following bugs were squashed: 9fabbab Disable hard-mute when a session has a media bug attached 0200bc1 FS-7083 Fix divide by zero [Jira: https://jira.freeswitch.org/browse/FS-7083] 067cb0f FS-7100 Make buffer for sub contact big enough in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7100] 7798b2f FS-6984 Set default video rates [Jira: https://jira.freeswitch.org/browse/FS-6984] 763e6aa FS-7046 Fix warning introduced from b341ff7 [Jira: https://jira.freeswitch.org/browse/FS-7046] 65e678b FS-7070 Fix mod_expr `clamp` function typo 0a66db6 FS-7111 Fix for bridge_early_media crash [Jira: https://jira.freeswitch.org/browse/FS-7111]

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freeswitch.org RSS feed problems
by Daniel Pocock 30 Dec '14

30 Dec '14

I've been using the RSS feed from FreeSWITCH.org at http://planet.sip5060.net Sometimes the planet script is getting a 500 error while fetching the feed https://freeswitch.org/feed/ Is anybody able to check the web server logs and see if it is running OK? Regards, Daniel

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No ringing is heard if carrier sends 180 Ringing - works fine when 183 Ringing (with SDP and RTP)
by Ali Pey 30 Dec '14

30 Dec '14

Hello, Here is the call scenario: - a-leg is bridged/connected to b-leg - They both are put on hold. a-leg listens to music, b-leg hears a prompt - A call is originated to c-leg: originate and park - If originate is successful. c-leg and b-leg are bridged together: uuid_bridge - b-leg does not hear ring back tone of c-leg of carrier send 180 Ringing, but would hear the Ring back tone if it's played from far end through RTP (183 Ringing) I have tried various things but haven't been able to make it work. b-leg hears dead air until c-leg actually answers. I can't not play any thing to b-leg either. Is there a way to make this work? Any suggestions? Thanks, Ali Pey

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FreeSwitch and WebRTC
by Adam Ben-Ayoun 30 Dec '14

30 Dec '14

Hi, We are using FreeSwitch as an MCU for audio-only WebRTC (basically, users connect to conference rooms). We have full control over the client application (it's a mobile app we are developing) and we solely rely on the client to choose the codec and do all the intelligent media related stuff. My question is, how can we minimize the audio processing/transcoding on FreeSwitch/mod_conference and mitigate the related overhead? Can we achieve that with tuning mod_conference's "rate" and "interval" parameters in that sense? Any other recommendations regarding FreeSwitch and WebRTC? Thanks, Adam

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Difference between 1.4 and 1.5 freeswitch version
by Shisheer Teli 30 Dec '14

30 Dec '14

Dear Team, Kindly share difference between freeswitch 1.4 and freeswitch 1.5 version. -- Regards, Shisheer T

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