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Send DTMF via ESL
by Gonzalo Gasca Meza 25 Jan '17

25 Jan '17

I'm using ESL to start calls by using originate command. Once a call is in ACTIVE state can we send DTMFs via ESL using uuid ? If so what would be the command? Thanks -Gonzalo

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Have you seen FreeSWITCH's Tips and Tricks?
by Kathleen King 25 Jan '17

25 Jan '17

Hello FreeSWITCHers! The FreeSWITCH team is working hard on getting version 1.8 ready for release. While you wait, we thought you might be interested in learning a few FreeSWITCH tricks you can try yourself. We had a look into our cookbook and found a few recipes you may be interested in. Have a look and don't forget you can learn even more about FreeSWITCH and other open source projects at ClueCon 2017 http://cluecon.com <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c3…> Don't miss our early bird pricing available for only a short time! *Simple Call Failover* How often do you miss a customer’s phone call? It always seems like the phone rings at the most inconvenient time and ends just before you can get to it. Wouldn’t it be great if instead of going to voicemail it just rang your co-worker? Well, FreeSWITCH has a solution for you. What if you didn’t have to answer those calls by yourself? In the FreeSWITCH dialplan you can configure something called Simple Call Failover. What this means is a call comes in and rings multiple endpoints (two separate desk phones in this example, one right after the other) and the person that answers is connected to the call. That way if you are busy someone else can answer that call and keep the customers happy. To set this up open up your dialplan, in this example we will use the default located at conf/dialplan/default.xml in a text editor in our terminal. Navigate to the extension you would like to use and add these lines: <extension name="failover_example"> <condition field="destination_number" expression="^(1000)$"> <action application="bridge" data="{ignore_early_media= true}sofia/internal/first@domain.com|sofia/internal/second@domain.com"/> </condition> </extension> By adding a pipe between your endpoints you are telling FreeSWITCH to dial those endpoints sequentially. The first one that is successfully connected will get the call and FreeSWITCH will cease dialing the additional endpoints. Now, to do this FreeSWITCH needs to ignore any early media like ringing so we added ignore_early_media=true at the beginning to tell FreeSWITCH to only connect the call when it is answered. It is as easy as that. *Time of Day Routing* Did you know you can configure FreeSWITCH to route to certain extensions based on the time of day? This is particularly convenient if you want to play a different greeting based on your office’s hours. To configure this you first need to define some variables for FreeSWITCH to use as a reference. For this example, we will assume our office is open 8:00 a.m. to 4:00 p.m. We start by adding this extension to the beginning of your dialplan context: <extension name="Time of day, day of week example" continue="true"> <condition wday="2-6" hour="8-15" break="never"> <action application="set" data="office_status=open" inline="true"/> <anti-action application="set" data="office_status=closed" inline="true"/> </condition> </extension> Now you can use the previously defined variables to set your extensions. In the example below FreeSWITCH will first check to see if the office is open or closed based on the first condition we defined. After that, FreeSWITCH will then execute whichever dialplan extension is necessary depending on the value of ‘office_status’ in the first condition. This configuration allows you to set specific actions depending on the time of day and what you would like your IVR to do. In this example we are initiating a particular IVR depending on whether the office is open or closed. <extension name="Office_Hours, 2001_X"> <condition field="destination_number" expression="^(2001)$"> <action application="execute_extension" data="2001_${office_status}"/> </condition> </extension> <extension name="Open"> <condition field="destination_number" expression="^(2001_open)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="500"/> </condition> </extension> <extension name="Closed"> <condition field="destination_number" expression="^(2001_closed)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="500"/> </condition> </extension> We have also added continue=”true” and break=”never” in our conditions. These tell FreeSWITCH to keep looking for more matches and without them FreeSWITCH would stop. Additionally, it should be noted that by setting variables we can easily allow for more conditions. If we were to set the “open” and “closed” extensions in an action tag and anti-action tag we are limited by only two options. In our current example you could go on to add things like lunch or holiday hours. *Manipulating SIP Request URIs* Another cool feature in FreeSWITCH is the ability to manipulate SIP To: Headers. You can use customized SIP To: Headers when routing calls to endpoints registered within your setup. This lets you send the DID in cases where a single registration is representing multiple DIDs. There isn’t a single flag to set in order to do this so you will have to configure your dialplan to grab the user’s registration information and parse out the username, replacing it with your custom setting. Be sure to only replace the username to prevent issues with things like NAT traversal. For this example, we have configured DID 123-456-7890, a username of user01, and domain of example.com. Our example extension will look like this: <extension name="call_1234567890"> <condition field="destination_number" expression="^(?:\+1|1)?1234567890$"/> <condition field="${sofia_contact(user01(a)example.com)}" expression="^[^\(a)]+(.*)"> <action application="bridge" data="sofia/external/123457890$1"/> </condition> </extension> This example shows the extension grabbing the user01 dial string and replacing the username with the preferred DID. In the first condition we are matching the determined DID and only if the incoming call matches that DID will the extension execute the actions. The second condition uses the results of an API, sofia_contact, to return the results as a value. This puts everything after the @ into the $1 variable which is then called during the bridge after our custom header. This preserves all the information required for contacting the server and sending the call to the correct destination. Be sure to register at https://cluecon.com <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c3…> for ClueCon 2017 to see all the latest and greatest the open-source community has to offer! <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c3…> For other tips like these, subscribe to our YouTube channel at https://youtube.com/freeswitch <http://freeswitch.us13.list-manage2.com/track/click?u=73f7dcee16e0a87ece76c…> or check for our books at https://freeswitch.org <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c3…> <http://freeswitch.us13.list-manage.com/track/click?u=73f7dcee16e0a87ece76c3…> <http://freeswitch.us13.list-manage2.com/track/click?u=73f7dcee16e0a87ece76c…> <http://freeswitch.us13.list-manage2.com/track/click?u=73f7dcee16e0a87ece76c…> *Copyright © 2017 FreeSWITCH Solutions, All rights reserved.* This is a mailing list for all things ClueCon! You have been subscribed to this list because you subscribed to the FreeSWITCH Users mailing list. If you would like to unsubscribe please follow the directions. *Our mailing address is:* FreeSWITCH Solutions 17345 Civic Drive #2531 Brookfield, WI 53045 Add us to your address book <http://freeswitch.us13.list-manage.com/vcard?u=73f7dcee16e0a87ece76c372d&id…> Want to change how you receive these emails? You can update your preferences <http://freeswitch.us13.list-manage.com/profile?u=73f7dcee16e0a87ece76c372d&…> or unsubscribe from this list <http://freeswitch.us13.list-manage.com/unsubscribe?u=73f7dcee16e0a87ece76c3…>

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Help us reach our goal! Support the FreeSWITCH project!
by Kathleen King 24 Jan '17

24 Jan '17

Hello FreeSWITCHers! Thank-you so much to those of you that have already made a contribution! We are almost halfway to our goal! Our developers are going to be hungry while they work this week and feeding the team is no easy chore. We are hoping to ease that burden with a little help from our community and your donations are really appreciated. And we all know a well-fed developer is a productive developer! Donate today to help the growing FreeSWITCH team stay focused on bringing you version 1.8.0 as soon as possible! <http://goog_1384205596/> https://www.gofundme.com/freeswitch-developer-meeting-2017 Kathleen King FreeSWITCH Public Relations Office: +1-213-286-0400 Mobile: +1-703-859-3757 http://freeswitch.org

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Fwd: NOTIFYs not received for conference SUBSCRIBE events
by Ram Anji 24 Jan '17

24 Jan '17

Hi, I am testing conference on freeswitch with jitsi. Jitsi sending SUBSCRIBE event after receiving 200 ok of call with focus parameter along with contact header. Server is responding 202 of subscription but no NOTIFY is observed for the subscription. I am testing with FreeSWITCH version: 1.6.14~64bit ( 64bit) where it is installed from freeswitch rpms and os is centos. Enclosed the console logs for the reference Regards, Raman

3 4

Bypass media
by Gregor Nanger 24 Jan '17

24 Jan '17

Hi! Need some advice. I would like to use FS for passing traffic. FS would accept calls from provider and terminate/bridge them to other provider. I was wondering if I can use bypass_media, so that RTP stream would not go through FS. Wouldthis would would work, since both providers uses IP authentication and on firewall only accepts traffic from/to allowed IPs. Or would RTP traffic go directly because of nature of SDP? Best regards, Gregor

2 2

Trunk for inbound connections
by Rick Jarvis 24 Jan '17

24 Jan '17

What is the best and cleanest method for providing a trunk for someone else to register into, in the most transparent way (i.e. cross-compatible, for providing a ITSP service to third party systems)?

3 6

mod_callcenter hitting 255 varchar limit on contact field in agent table using sofia_contact
by Shaun Stokes 24 Jan '17

24 Jan '17

Hi All, We are using sofia_contact to send calls to agents in mod_callcenter and we are hitting a 255 character limit on the agent contact with-in the FreeSWITCH backend DB. Unfortunately there's no way for us to reduce the string used for the agent contact since we need to include arguments for fail_on_single_reject, pickup and a few variables which otherwise wouldn't get passed. We use multiple registrations per extension, currently sofia_contact is the only way for us to correctly route to all registrations simultaneously with fail_on_single_reject. We've resolved this in our environment using the following process. sudo service freeswitch stop #### Modify the File: /freeswitch/mod/mod_callcenter.so ## HEX Address 00012250 -> # Replace: ANSI: 255 Hex Col 7: 32 Col 8: 35 Col 9: 35 # With: ANSI: 510 HEX Col 7: 35 Col 8: 31 Col 9: 30 The full ANSI line is (if the address is different): contact VARCHAR(255), To be replaced with: contact VARCHAR(510), #### Delete callcenter.db Drop Agents Table sudo service freeswitch start sudo fs_cli reload mod_callcenter I'd like to propose the 'contact' varchar field is increased to 510 for the 'agent' table in the source code 'mod_callcenter.so', or is there a better way of doing this? 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. ______________________________________________________________________

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leg-B cuts off connection while transfer
by Paul Zillmann 24 Jan '17

24 Jan '17

Hello Folks, a user hooked me on, that whenever he calls certain numbers, and was put into hold (call is transfered), that legB sends a BYE after all. Is this a misconfigure on the user's side - or is legB misconfigured? The siptrace attatched seems to be okay - but what I don't understand is, why is the second reinvite marked as recvonly. I guess the second reinvited comes from legB because of the transfer was accepted - but why reconly - and why does legB sends a BYE after that one? Thanks a lot ------------------------------------------------------------------------ recv 1075 bytes from udp/[10.0.0.2]:5060 at 16:45:47.854134: ------------------------------------------------------------------------ INVITE sip:022029****@172.16.10.11 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport From: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 To: <sip:022029****@172.16.10.11> Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 21 INVITE Contact: <sip:204@10.0.0.2:5060> Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK Max-Forwards: 70 P-Early-Media: supported User-Agent: SIPPER for PhonerLite Session-Expires: 1800 Supported: 100rel, replaces, from-change, timer P-Preferred-Identity: <sip:204@172.16.10.11> Content-Length: 406 v=0 o=- 2104407805 1 IN IP4 10.0.0.2 s=SIPPER for PhonerLite c=IN IP4 10.0.0.2 t=0 0 m=audio 5062 RTP/AVP 9 8 0 2 3 97 110 111 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1205340143 a=sendrecv ------------------------------------------------------------------------ send 325 bytes to udp/[10.0.0.2]:5060 at 16:45:47.854518: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060 From: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 To: <sip:022029****@172.16.10.11> Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 21 INVITE User-Agent: H6G Callserver Content-Length: 0 ------------------------------------------------------------------------ send 1417 bytes to udp/[217.0.23.100]:5060 at 16:45:47.911364: ------------------------------------------------------------------------ INVITE sip:022029****@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK782XNNX3Fm2Ur Route: <sip:217.0.23.100:5060;transport=udp;lr> Max-Forwards: 69 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N To: <sip:022029****@tel.t-online.de> Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127509 INVITE Contact: <sip:gw+P@172.16.10.11:5080;transport=udp;gw=P> User-Agent: H6G Callserver Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 360 P-Early-Media: supported X-FS-Support: update_display,send_info P-Asserted-Identity: "PhonerLite" <sip:0220224****@tel.t-online.de> P-Preferred-Identity: <sip:0220224****@tel.t-online.de> v=0 o=FreeSWITCH 1484904625 1484904626 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 22522 RTP/AVP 8 0 110 111 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:111 SPEEX/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:102 telephone-event/16000 a=fmtp:102 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 528 bytes from udp/[217.0.23.100]:5060 at 16:45:48.072425: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required 02035034C Via: SIP/2.0/UDP 172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK782XNNX3Fm2Ur To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_97483a920cc4ce8660c6c8f7126c6e9d From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127509 INVITE Content-Length: 0 Proxy-Authenticate: Digest nonce="CFFDD4D3B7308258000000003356C90F",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true ------------------------------------------------------------------------ send 426 bytes to udp/[217.0.23.100]:5060 at 16:45:48.072612: ------------------------------------------------------------------------ ACK sip:022029****@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK782XNNX3Fm2Ur Route: <sip:217.0.23.100:5060;transport=udp;lr> Max-Forwards: 69 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_97483a920cc4ce8660c6c8f7126c6e9d Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127509 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1702 bytes to udp/[217.0.23.100]:5060 at 16:45:48.073041: ------------------------------------------------------------------------ INVITE sip:022029****@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK8HvpQge7cXrem Route: <sip:217.0.23.100:5060;transport=udp;lr> Max-Forwards: 69 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N To: <sip:022029****@tel.t-online.de> Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 INVITE Contact: <sip:gw+P@172.16.10.11:5080;transport=udp;gw=P> User-Agent: H6G Callserver Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authorization: Digest username="jbvp-scharf(a)t-online.de", realm="tel.t-online.de", nonce="CFFDD4D3B7308258000000003356C90F", cnonce="Wqe3FFnKEjWDEkCNXBvrUw", algorithm=MD5, uri="sip:022029****@tel.t-online.de", response="f9665e5312227392a11b97914e11593d", qop=auth, nc=00000001 Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 360 P-Early-Media: supported X-FS-Support: update_display,send_info P-Asserted-Identity: "PhonerLite" <sip:0220224****@tel.t-online.de> P-Preferred-Identity: <sip:0220224****@tel.t-online.de> v=0 o=FreeSWITCH 1484904625 1484904626 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 22522 RTP/AVP 8 0 110 111 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:111 SPEEX/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:102 telephone-event/16000 a=fmtp:102 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 320 bytes from udp/[217.0.23.100]:5060 at 16:45:48.304747: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem To: <sip:022029****@tel.t-online.de> From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 876 bytes from udp/[217.0.23.100]:5060 at 16:45:49.434461: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 INVITE Contact: <sip:sgc_c@217.0.23.100;transport=udp> Record-Route: <sip:217.0.23.100;transport=udp;lr> P-Early-Media: sendonly Supported: timer Content-Type: application/sdp Content-Length: 225 Allow: NOTIFY, UPDATE, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER v=0 o=- 1144933594 2336235026 IN IP4 217.0.23.100 s=Basic Session c=IN IP4 217.0.4.167 t=0 0 m=audio 10224 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 1129 bytes to udp/[10.0.0.2]:5060 at 16:45:49.457465: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060 From: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 21 INVITE Contact: <sip:022029****@172.16.10.11:5060;transport=udp> User-Agent: H6G Callserver Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 220 P-Early-Media: sendonly P-Asserted-Identity: "[P] 022029****" <sip:022029****@172.16.10.11> v=0 o=FreeSWITCH 1484896495 1484896496 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 30654 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 867 bytes from udp/[217.0.23.100]:5060 at 16:45:49.838384: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 INVITE Contact: <sip:sgc_c@217.0.23.100;transport=udp> Record-Route: <sip:217.0.23.100;transport=udp;lr> P-Early-Media: sendonly Supported: timer Content-Type: application/sdp Content-Length: 225 Allow: NOTIFY, UPDATE, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER v=0 o=- 1144933594 2336235026 IN IP4 217.0.23.100 s=Basic Session c=IN IP4 217.0.4.167 t=0 0 m=audio 10224 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ recv 1132 bytes from udp/[217.0.23.100]:5060 at 16:46:02.806318: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 INVITE Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Record-Route: <sip:217.0.23.100;transport=udp;lr> Session-Expires: 1800;refresher=uas Supported: timer Content-Type: application/sdp Content-Length: 225 Session-ID: cf2f265e180b6a914ef96ffa72c934f5 Authentication-Info: qop=auth,rspauth="91a1f6e74a09e15ea0682500b4dc281f",cnonce="Wqe3FFnKEjWDEkCNXBvrUw",nc=00000001 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE v=0 o=- 1144933594 2336235026 IN IP4 217.0.23.100 s=Basic Session c=IN IP4 217.0.4.167 t=0 0 m=audio 10224 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 794 bytes to udp/[217.0.23.100]:5060 at 16:46:02.808824: ------------------------------------------------------------------------ ACK sip:sgc_c@217.0.23.100;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK9tNFSBZaa6e1F Route: <sip:217.0.23.100;transport=udp;lr> Max-Forwards: 70 From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N To: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127510 ACK Contact: <sip:gw+P@172.16.10.11:5080;transport=udp;gw=P> Proxy-Authorization: Digest username="jbvp-scharf(a)t-online.de", realm="tel.t-online.de", nonce="CFFDD4D3B7308258000000003356C90F", cnonce="Wqe3FFnKEjWDEkCNXBvrUw", algorithm=MD5, uri="sip:022029****@tel.t-online.de", response="f9665e5312227392a11b97914e11593d", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 1118 bytes to udp/[10.0.0.2]:5060 at 16:46:02.826970: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060 From: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 21 INVITE Contact: <sip:022029****@172.16.10.11:5060;transport=udp> User-Agent: H6G Callserver Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 220 P-Asserted-Identity: "[P] 022029****" <sip:022029****@172.16.10.11> v=0 o=FreeSWITCH 1484896495 1484896496 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 30654 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 399 bytes from udp/[10.0.0.2]:5060 at 16:46:02.859203: ------------------------------------------------------------------------ ACK sip:022029****@172.16.10.11:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK80af5b3795dde611a221b1b68e4749c1;rport From: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 21 ACK Contact: <sip:204@10.0.0.2:5060> Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1011 bytes from udp/[217.0.23.100]:5060 at 16:46:36.717056: ------------------------------------------------------------------------ INVITE sip:gw+P@172.16.10.11:5080;transport=udp;gw=P SIP/2.0 Max-Forwards: 66 Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127511 INVITE Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Min-Se: 900 Session-Expires: 1800;refresher=uac Supported: timer Supported: 100rel Content-Type: application/sdp Content-Length: 237 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE v=0 o=- 1144933594 2336235027 IN IP4 217.0.23.100 s=Basic Session c=IN IP4 217.0.4.167 t=0 0 a=sendonly m=audio 10224 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendonly ------------------------------------------------------------------------ send 404 bytes to udp/[217.0.23.100]:5060 at 16:46:36.717365: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127511 INVITE User-Agent: H6G Callserver Content-Length: 0 ------------------------------------------------------------------------ send 975 bytes to udp/[217.0.23.100]:5060 at 16:46:36.969711: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127511 INVITE Contact: <sip:gw+P@172.16.10.11:5080;transport=udp;gw=P> User-Agent: H6G Callserver Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 232 v=0 o=FreeSWITCH 1484904625 1484904627 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 22522 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=inactive a=ptime:20 ------------------------------------------------------------------------ recv 540 bytes from udp/[217.0.23.100]:5060 at 16:46:37.038798: ------------------------------------------------------------------------ ACK sip:gw+P@172.16.10.11:5080;transport=udp;gw=P SIP/2.0 Max-Forwards: 66 Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ij54ucc23me1yvy9vdmn4y96yh To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127511 ACK Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Content-Length: 0 ------------------------------------------------------------------------ recv 1011 bytes from udp/[217.0.23.100]:5060 at 16:46:49.474526: ------------------------------------------------------------------------ INVITE sip:gw+P@172.16.10.11:5080;transport=udp;gw=P SIP/2.0 Max-Forwards: 66 Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127512 INVITE Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Min-Se: 900 Session-Expires: 1800;refresher=uac Supported: timer Supported: 100rel Content-Type: application/sdp Content-Length: 237 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE v=0 o=- 1144933594 2336235028 IN IP4 217.0.23.100 s=Basic Session c=IN IP4 217.0.4.167 t=0 0 a=recvonly m=audio 10224 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=ptime:20 a=recvonly ------------------------------------------------------------------------ send 404 bytes to udp/[217.0.23.100]:5060 at 16:46:49.474824: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3 From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127512 INVITE User-Agent: H6G Callserver Content-Length: 0 ------------------------------------------------------------------------ send 963 bytes to udp/[217.0.23.100]:5060 at 16:46:49.767507: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3 From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127512 INVITE Contact: <sip:gw+P@172.16.10.11:5080;transport=udp;gw=P> User-Agent: H6G Callserver Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 220 v=0 o=FreeSWITCH 1484904625 1484904628 IN IP4 172.16.10.11 s=FreeSWITCH c=IN IP4 172.16.10.11 t=0 0 m=audio 22522 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 540 bytes from udp/[217.0.23.100]:5060 at 16:46:49.837652: ------------------------------------------------------------------------ ACK sip:gw+P@172.16.10.11:5080;transport=udp;gw=P SIP/2.0 Max-Forwards: 66 Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ishs6rygvv7pvgg3qpcfprg24g To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127512 ACK Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Content-Length: 0 ------------------------------------------------------------------------ recv 577 bytes from udp/[217.0.23.100]:5060 at 16:47:03.694027: ------------------------------------------------------------------------ BYE sip:gw+P@172.16.10.11:5080;transport=udp;gw=P SIP/2.0 Max-Forwards: 66 Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i241fgoep8n7wg8m5jfzdz6q59 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127513 BYE Reason: Q.850;cause=16;eri-location=0 Content-Length: 0 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE ------------------------------------------------------------------------ send 541 bytes to udp/[217.0.23.100]:5060 at 16:47:03.706495: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i241fgoep8n7wg8m5jfzdz6q59 From: <sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466 To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53 CSeq: 102127513 BYE User-Agent: H6G Callserver Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 538 bytes to udp/[10.0.0.2]:5060 at 16:47:03.709373: ------------------------------------------------------------------------ BYE sip:204@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.11;rport;branch=z9hG4bK7rN6peF2S5pUB Max-Forwards: 70 From: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB To: "PhonerLite" <sip:204@172.16.10.11>;tag=2258694084 Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1(a)10.0.0.2 CSeq: 102127547 BYE User-Agent: H6G Callserver Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;eri-location=0 Content-Length: 0

2 4

mod_radius_cdr
by abc1 24 Jan '17

24 Jan '17

Hi List Is there a DB schema for generating the radacct table in freeradius to work with mod_radius_cdr? Thanks! Alex

2 1

sip_invite_params for all outgoing calls
by René Weiss 23 Jan '17

23 Jan '17

Hi, Is it possible to set "sip_invite_params='user=phone'" for all outgoing calls / all outgoing calls using a specific gateway? I'm searching a solution which work with the dialplan and also with calls over mod_event_socket and "originate". I tried to set the variable in "conf/sip_profiles/external.xml" and "conf/sip_profiles/external/outgoing.xml" but I don't know if this was the wrong place or if I was just using the wrong syntax. Thanks, René

2 6