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Freeswitch registration error with GSM gateway
by Deepika Yadav 05 Jul '17

05 Jul '17

Hi, I have a VOIP-GSM 16 port Dinstar gateway installed. It has its local IP, SIM inserted and receiving signals but somehow it is not getting registered with Freeswitch as external profile. I have tried all the things found at different sources such as : https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo> http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo> but no luck :( *Error at console* - Failed Registration with status Request Timeout [408] *SIP trace is as follows*: Freeswitch Server IP - *192.168.2.71* Gateway IP - *192.1.68.2.69* SUBSCRIBE sip:dwg2008@192.168.2.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.69;branch=z9hG4bKf22aec329cf8392a03d1fb7ac5907373;rport From: <sip:dwg2008@192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907 To: <sip:dwg2008@192.168.2.71> Call-ID: 7901887bd57699805a51f44e8e162cd5(a)192.168.2.69 CSeq: 479122789 SUBSCRIBE Contact: <sip:dwg2008@192.168.2.69> Event: message-summary Supported: eventlist Expires: 1 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER Max-Forwards: 70 Accept: application/simple-message-summary Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN From: <sip:dwg2008@192.168.2.71>;tag=5umvaME1a8N6 To: <sip:dwg2008@192.168.2.71>;tag=8b79a39dab9333d980ae172b0ddf0b10 Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223(a)192.168.2.69 CSeq: 797846101 NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg From: <sip:dwg2008@192.168.2.71>;tag=DaCYUMXLeOft To: <sip:dwg2008@192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907 Call-ID: 7901887bd57699805a51f44e8e162cd5(a)192.168.2.69 CSeq: 797846103 NOTIFY Content-Length: 0 REGISTER sip:192.168.2.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.69:5060 ;branch=z9hG4bKac7ee3cb94dbc6021702a40d789e140c;rport From: <sip:dwg2008@192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1 To: <sip:dwg2008@192.168.2.71> Call-ID: f58080d7fd4106bc3673e1295108eb09(a)192.168.2.69 CSeq: 479121941 REGISTER Contact: <sip:dwg2008@192.168.2.69:5060> Expires: 1800 User-Agent: DWG2000F Business 20231220 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER Max-Forwards: 70 Content-Length: 0 REGISTER sip:192.168.2.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.69:5060 ;branch=z9hG4bKc2d768329e8115ca788edb78700b7db5;rport From: <sip:dwg2008@192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1 To: <sip:dwg2008@192.168.2.71> Call-ID: f58080d7fd4106bc3673e1295108eb09(a)192.168.2.69 CSeq: 479121942 REGISTER Contact: <sip:dwg2008@192.168.2.69:5060> Authorization: Digest username="dwg2008", realm="192.168.2.71", nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f", uri="sip:192.168.2.71", response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5, cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001 Expires: 1800 User-Agent: DWG2000F Business 20231220 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER Max-Forwards: 70 Content-Length: 0 -- Regards Deepika https://www.iiitd.edu.in/~deepikay/

2 4

Music on hold
by Gregor Nanger 04 Jul '17

04 Jul '17

Need some advice. I have multi tenant setup, driven with xml_curl. What is best approach to make music on hold configurable for every customer? I mean from resource point of view. Music files should be read from http, not file system. Thank you, Gregor

3 4

Limit
by Shaun Stokes 04 Jul '17

04 Jul '17

Hi All, In our environment we restrict the number of current channels (using limit) per extension, however using attended transfer allows extensions to exceed the limit since leg a ends once the transfer completes which resets the limit to 0. I believe the solution is to apply the limit on leg b, however leg b is initiated via a bridge (with-in our outbound LUA script). How can we apply the limit to leg b, or is there a better solution? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

4 5

mod_rad_auth dial plan condition to match radius result
by Jospeh Waite 04 Jul '17

04 Jul '17

Hi Guys Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected. I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made. I have put the following in the dial plan <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 credit_time=${credit_time}"/> <action application="log" data="INFO return_code=${return_code}"/> <condition field="${return_code}" expression="^16$"> <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> And I check the freeswitch log and see EXECUTE sofia/external/07966677711(a)185.8.92.3 log(INFO AUTH_RESULT=OK) 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 AUTH_RESULT=OK EXECUTE sofia/external/07966677711(a)185.8.92.3 log(INFO credit_time=7199) 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 credit_time=7199 EXECUTE sofia/external/07966677711(a)185.8.92.3 log(INFO return_code=16) 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 return_code=16 But it fails to execute the bit within the condition. Any Ideas of how to make this work? Regards

4 7

Need channel variables stored in DB in realtime
by Sagar Malam 04 Jul '17

04 Jul '17

Hello, I want to have all the channel variables stored in DB in realtime.I tried making ODBC configuration for MySQL for FS core data but i did not find any table with channel variable information. Is there any inbuilt way to do so ? Please help. Thanks in advance -- Thanks & Regards, Sagar Malam Team Lead | Ecosmob Technologies Pvt. Ltd. (+91)9601533171 | www.ecosmob.com <http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=A…> Skype: sagar.ecosmob

4 3

Max transmissions
by Rick Jarvis 03 Jul '17

03 Jul '17

Where should I start looking to fault find this? I’m getting choppy audio (FS->External) at the start of a call, which I’m guessing might be connected? 2017-07-03 16:51:00.631873 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53538 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:00.831873 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53539 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:01.031875 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53540 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:01.231873 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53541 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:01.431886 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53542 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:01.631874 [DEBUG] switch_rtp.c:1464 [ zrtp utils]: Send <HELLO> ssrc=2371749770 seq=53543 size=148. Stream 2:CLEAR:START 2017-07-03 16:51:01.831873 [DEBUG] switch_rtp.c:1464 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=2 2017-07-03 16:51:01.831873 [DEBUG] switch_rtp.c:1464 [ zrtp]: Stream ID=2 CLEAR switching <START> ---> <NOZRTP>.

1 0

Making DTMF tone 'silent' to caller
by Mark Melling 03 Jul '17

03 Jul '17

Hi, I have a scenario where a mobile phone makes a 'normal' call (not SIP/VoIP) into a Freeswitch conference (via a SIP trunk). The dialled number includes DTMF tones to be transmitted after the call is dialled. When sent these DTMF tones can be heard by the mobile phone user. What I want to achieve is that these tones are not audible to the mobile caller. In a previous incarnation of this system the mobile phone dialled into a Twilio hosted conference. In the set-up when the DTMF tones were transmitted the mobile phone user could not hear them. Is it the case that Freeswitch is generating the DTMF tones and if so how can I stop this? If not, what are the possible explanations - e.g. perhaps DTMF tones are played in-band in one scenario versus out-of-band in the other? Any suggestions and thoughts most appreciated. Thanks Mark

1 0

000298C appends to the destination number
by Artur Mega 02 Jul '17

02 Jul '17

Hello, anybody knows why destination number starts with 000298C (in several cases)? I think it is encoded plus sign. -- Regards, Arthur

1 0

FreeSwitch base docker container
by Sergey Safarov 02 Jul '17

02 Jul '17

Hello I created minimized version of could you look pull request 1322 <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1322> or test docker container safarov/freeswitch/ <https://hub.docker.com/r/safarov/freeswitch/> I ask FreeSwitch core team to consider the possibility of making it an official container on the Docker Hub Sergey afarov

1 0

Load testing
by Siju Nair 02 Jul '17

02 Jul '17

Hi team, Any approximate figure regarding FS Server load balance? How much concurrent calls server can handle also minimum hardware configuration that required to handle 500 concurrent calls at a time ? Sent from my iPhone > On 05-Apr-2017, at 9:18 AM, Siju Nair <siju.irs(a)gmail.com> wrote: > > Hi Micheal > > Thank you very much for ur reply. > Can I know what all info u are looking for I am using FS version 1.2.23 > > Am running call center module in FS with 4 agents taking calls .. for 3 days it worked fine but then it get hanged up and I was unable to log in to server .. > > I could see above screenshot logs . > > Sent from my iPhone > >> On 05-Apr-2017, at 1:29 AM, Michael Jerris <mike(a)jerris.com> wrote: >> >> We’ve looked at the provided information. This information is not enough to provide any feedback on this issue. >> >>> On Apr 3, 2017, at 11:34 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>> >>> Hi team >>> Any advice on below issue, please!! >>> >>> Sent from my iPhone >>> >>>> On 03-Apr-2017, at 1:00 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>>> >>>> Hi team, >>>> Please help me on below issue !! >>>> >>>> Sent from my iPhone >>>> >>>>> On 09-Mar-2017, at 4:44 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>>>> >>>>> Hi team, >>>>> >>>>> Please help on below issue ! Also my CPU avg is 42 on calls ! >>>>> >>>>> Sent from my iPhone >>>>> >>>>>> On 08-Mar-2017, at 7:53 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>>>>> >>>>>> Please check the attached screen shot for more clarification >>>>>> >>>>>>> On Wed, Mar 8, 2017 at 7:48 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>>>>>> Hi team, >>>>>>> >>>>>>> My FS server got hung, under server logs due freeswitch server got >>>>>>> hung up. Below is the log i found.. >>>>>>> >>>>>>> echo 0 > /proc/sys/kernal/hung_task_timeout_secs"disables this message >>>>>>> info: task freeswitch: 3034 blocked for more than 120 secs. >>>>>>> >>>>>>> and then >>>>>>> freeswitch > waiting for background process to be ready. >>>>>>> >>>>>>> Please help me on the same. >>>>>>> >>>>>>> Sent from my iPhone >>>>>> >>>>>> <ccmmydqld_hang.PNG> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)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(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)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(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org

1 0