300 Multiple Choices - how to force FS to replace From field
Hello everyone, I would like to express my thanks in advance to anyone who may be able to help me with some insides. I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process. 1. Sending call to FS with Zoiper, destination 556230951662 INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK-d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2. FS sending INVITE to SIP Redirect server INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556) SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server 4. FS will send the call to the Termination IP WITHOUT changing the FROM field INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 This is the dialplan I use to send calls to SIP redirect server <include> <extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension> <extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension> <extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/> <condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call-origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_number=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@ ${sip_network_ip}}sofia/internal/${destination_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated! Best regards, L. Popov
Why are you using the from field for authentication? On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello everyone,
I would like to express my thanks in advance to anyone who may be able to help me with some insides.
I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process.
1. Sending call to FS with Zoiper, destination 556230951662
INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232
v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv
2. FS sending INVITE to SIP Redirect server
INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party= calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556)
SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server
4. FS will send the call to the Termination IP WITHOUT changing the FROM field
INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party= calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
This is the dialplan I use to send calls to SIP redirect server
<include>
<extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension>
<extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension>
<extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/>
<condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call- origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_ name=${sip_req_user}"/> <action application="set" data="origination_caller_id_ number=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri= sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${ destination_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include
Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated!
Best regards,
L. Popov
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Hello Brian, The authentication is DIGEST done via RADIUS. The Username is the same as the Caller ID...or maybe I understood your question wrong..? The number that shows in the from field is actually the SIP username created in the system ( for routing, billing, radius AAA, etc. ) and it is as well the callers number ( Caller ID). I use Raduis AAA to authenticate and account the calls and the user you see 551000 is actually a username of voip account created in the billing. That is why you see the incoming call from that username (551000). This is what FS is using in the FROM field. Since many of the accounts are created in format different from E164, often it is necessary to rewrite the account number ( the caller number ) to E164 or the termination will not accept the call. This is what I am trying to do now, set a rewrite rule in the billing system to convert 551000 to 1140031556 and this is what it is returned to FS as you can see in the packets the billing sends back.. Cheers, L.Popov On Wed, May 3, 2017 at 7:20 PM, Brian West <brian@freeswitch.org> wrote:
Why are you using the from field for authentication?
On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello everyone,
I would like to express my thanks in advance to anyone who may be able to help me with some insides.
I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process.
1. Sending call to FS with Zoiper, destination 556230951662
INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232
v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv
2. FS sending INVITE to SIP Redirect server
INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556)
SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server
4. FS will send the call to the Termination IP WITHOUT changing the FROM field
INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
This is the dialplan I use to send calls to SIP redirect server
<include>
<extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension>
<extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension>
<extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/>
<condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call-origin =originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_na me=${sip_req_user}"/> <action application="set" data="origination_caller_id_nu mber=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina tion_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include
Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated!
Best regards,
L. Popov
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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--
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*Twitter: @FreeSWITCH , @briankwest*
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*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
are you setting manual-redirect in your Sofia profile? https://freeswitch.org/confluence/display/FREESWITCH/Handling+SIP+Redirect On Thu, May 4, 2017 at 10:20 AM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello Brian,
The authentication is DIGEST done via RADIUS. The Username is the same as the Caller ID...or maybe I understood your question wrong..? The number that shows in the from field is actually the SIP username created in the system ( for routing, billing, radius AAA, etc. ) and it is as well the callers number ( Caller ID). I use Raduis AAA to authenticate and account the calls and the user you see 551000 is actually a username of voip account created in the billing. That is why you see the incoming call from that username (551000). This is what FS is using in the FROM field. Since many of the accounts are created in format different from E164, often it is necessary to rewrite the account number ( the caller number ) to E164 or the termination will not accept the call. This is what I am trying to do now, set a rewrite rule in the billing system to convert 551000 to 1140031556 and this is what it is returned to FS as you can see in the packets the billing sends back..
Cheers, L.Popov
On Wed, May 3, 2017 at 7:20 PM, Brian West <brian@freeswitch.org> wrote:
Why are you using the from field for authentication?
On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello everyone,
I would like to express my thanks in advance to anyone who may be able to help me with some insides.
I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process.
1. Sending call to FS with Zoiper, destination 556230951662
INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232
v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv
2. FS sending INVITE to SIP Redirect server
INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556)
SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server
4. FS will send the call to the Termination IP WITHOUT changing the FROM field
INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
This is the dialplan I use to send calls to SIP redirect server
<include>
<extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension>
<extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension>
<extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/>
<condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call-origin =originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_na me=${sip_req_user}"/> <action application="set" data="origination_caller_id_nu mber=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina tion_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include
Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated!
Best regards,
L. Popov
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST) <https://freeswitch.com/appointment>
Allison prompts for FreeSWITCH:
*https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch>
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) <https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest
Hello Brian, Sorry, did not see your response on May, 5th. Yes I have this in sofia profile. <param name="manual-redirect" value="true"/> Is there anything else that I need to add or change in order to make FS to use the returned FROM value by the SIP REDIRECT server? Cheers, L.Popov On Fri, May 5, 2017 at 10:31 AM, Brian West <brian@freeswitch.org> wrote:
are you setting manual-redirect in your Sofia profile?
https://freeswitch.org/confluence/display/FREESWITCH/Handling+SIP+Redirect
On Thu, May 4, 2017 at 10:20 AM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello Brian,
The authentication is DIGEST done via RADIUS. The Username is the same as the Caller ID...or maybe I understood your question wrong..? The number that shows in the from field is actually the SIP username created in the system ( for routing, billing, radius AAA, etc. ) and it is as well the callers number ( Caller ID). I use Raduis AAA to authenticate and account the calls and the user you see 551000 is actually a username of voip account created in the billing. That is why you see the incoming call from that username (551000). This is what FS is using in the FROM field. Since many of the accounts are created in format different from E164, often it is necessary to rewrite the account number ( the caller number ) to E164 or the termination will not accept the call. This is what I am trying to do now, set a rewrite rule in the billing system to convert 551000 to 1140031556 and this is what it is returned to FS as you can see in the packets the billing sends back..
Cheers, L.Popov
On Wed, May 3, 2017 at 7:20 PM, Brian West <brian@freeswitch.org> wrote:
Why are you using the from field for authentication?
On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov <koki.roul@gmail.com> wrote:
Hello everyone,
I would like to express my thanks in advance to anyone who may be able to help me with some insides.
I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process.
1. Sending call to FS with Zoiper, destination 556230951662
INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232
v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv
2. FS sending INVITE to SIP Redirect server
INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556)
SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server
4. FS will send the call to the Termination IP WITHOUT changing the FROM field
INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=c alling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30
This is the dialplan I use to send calls to SIP redirect server
<include>
<extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension>
<extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension>
<extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/>
<condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call-origin =originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_na me=${sip_req_user}"/> <action application="set" data="origination_caller_id_nu mber=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina tion_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/
</condition> </extension> </include
Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated!
Best regards,
L. Popov
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST) <https://freeswitch.com/appointment>
Allison prompts for FreeSWITCH:
*https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch>
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST) <https://freeswitch.com/appointment>
Allison prompts for FreeSWITCH:
*https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch>
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Atenciosamente, ============================ Lyubo Popov CEO - BlasterPhone LLC Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556) iNum: +883 510001-354111 Website: http://www.blastervoip.com.br/ ============================
participants (3)
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Brian West
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Lyubo Popov
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Lyubo Popov