search for: 100rel

Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non provisional response) therefore the calling server does not send a PRACK. I have the same config at both ends and if I change the 1...

hello I want to know how to pass through 100rel header. and I hope that asterisk PRACK to UAS.(RFC3262 behavior) _________________________________________________________________

...th a port forwarding (5060). This configuration has already been validated with another VoIP provider, but in the present case, not. I suppose (thanks to the sip trace) my asterisk is not able to answer a call which need Prack. My asterisk answer : ' SIP/2.0 420 Bad extension ... Unsupported: 100rel ' Any idea ? It is because I use port forwarding ? Should i have to open other port ? Thanks -- Jean-Baptiste Bellet Ing?nieur D?veloppement Lucyde SAS Prologue 1 - La Pyr?n?enne BP 27201 LABEGE cedex +33 (0)5 34 31 86 36 http://www.lucyde.com

...168.45.129>;tag=812da8c0-13c4-46277c06-279cd106-42ff Call-ID: 3ee92dbe77f51a1748f736be4593719d@161.49.142.250 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Content-Length: 0 <-------------> ?--- (10 headers 0 lines) --- ? [Khfemsrv*CLI> Really destroying SIP dialog '3ee92dbe77f51a1748f736be4593719d@161.49.142.250' Method: OPTIONS ? [Khfemsrv*CLI>...

...as02fa8fcc To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE Server: FPBX-13.0.188.8(13.11.2) Contact: <sip:80.142.13.32:55060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 286 v=0 o=- 15363811 15363814 IN IP4 192.168.3.99 s=Asterisk c=IN IP4 80.142.13.32 t=0 0 m=audio 51822 RTP/AVP 8 3 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/80...

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and PRACK. t38 is tested and working fine with Zoiper client but I can't get the t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found FAXCOM announces that it supports 100rel so I added the PRACK patch hoping that would do the trick. Now it gets a little further but * complains about rejecting a non-primary audio stream. Could this be a problem with 1.8 not liking the second media stream or is there some more configuration tweaking to be done? --- CUT ----------------...

...p:+91712442211 at unknown.invalid>. Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>. From: <sip:+91711189078 at 10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602. P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>. Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: <sip:+91712442211 at unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211 at unknown.invalid;user=phone>;index=1.1. Max-Forwards: 69....

...4964319" <sip:4414964319@10.70.0.92>;tag=as75a2b003 To: <sip:2924357@pbx-quantum>;tag=SD6q75199-quantum1.quantum.bm+1+acb61+6a3a58ac Call-ID: 296c97ab6b5f1b2444d51cf9681a7c42@10.70.0.92 CSeq: 102 INVITE Server: DC-SIP/2.0 Allow-Events: message-summary Allow-Events: refer Supported: 100rel Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REGISTER Allow: OPTIONS Allow: PRACK Allow: UPDATE Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Accept-Encoding: identity Accept: application/sdp Accept: application/simple-message-summary Contact: <sip:2924357@127.0.0.100:5060;maddr=10.70....

...59458 INVITE Content-Type: application/sdp Contact: <sip:3x8x6x3x3x at xx.yy.xx.yy:5060;user=phone;transport=udp> User-Agent: Nortel SESM 14.1.0.12 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel Remote-Party-ID: <sip:3x8x6x3x3x at sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL P-Asserted-Identity: <sip:3x8x6x3x3x at 10.39.1.19;user=phone> Allow: UPDATE,REFER Content-Length: 293 v=0 o=- 0 138163748 IN IP4 xx.yy.xx.yy s...

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

...Hector" <sip:201 at 10.1.0.10>;tag=uucwz Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org CSeq: 183 INVITE Contact: <sip:201 at 10.1.0.65> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 2122879389 441437466 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------&...

...d: INFO sip:5854199708@208.34.86.35;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacyRPrQQu From:<sip:1017@m2k.netacc.net>;tag=1c11546 To: <sip:5854199708@208.34.86.35>;tag=as37f2b147 Call-ID: 2879790839083xQxj-1017--5854199708@208.34.86.37 CSeq: 3100834 INFO Supported: 100rel,em Content-Type: application/sdp Content-Length: 35 p=Digit-Collection y=Digits d=7 9 headers, 3 lines Receiving DTMF! Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacyRPrQQu From: <sip:1017@m2k.netacc.net>;tag=1c11546 To: <sip:5854199708@208.34.86.35&...

...10.1.1.1:5060;branch=z9hG4bK-29450-3-0 Max-Forwards: 60 From: <sip:39937841 at 192.168.225.2:5061;user=phone>;tag=3 To: <sip:039988120F at 172.16.60.160:5060;user=phone> Call-ID: 3-29450 at 172.16.60.160 CSeq: 1 INVITE Contact: <sip:10.1.1.1:5060> User-Agent: Simulator Supported: 100rel Privacy: id Min-SE: 90 Content-Type: application/sdp Content-Length: 201 v=0 o=172.16.60.160 10864 2 IN IP4 172.16.60.160 s=SIP Call c=IN IP4 172.16.60.160 t=0 0 a=sendrecv m=audio 60000 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <--- Transmitting SIP respon...

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

...2797887@172.16.84.100;to-tag=2216503390;from-tag=5e7xzl7sqh P-Key-Flags: keys="3" User-Agent: snom200-3.42 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 292 v=0 o=root 510805774 510805774 IN IP4 4.12.220.193 s=call c=IN IP4 4.12.220.193 t=0 0 m=audio 10002 RTP/AVP 0 8 3 18 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:...

...4> > Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060> > Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3 > CSeq: 6753 INVITE > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, > UPDATE, PRACK, REGISTER, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > in-SE: 90 > Content-Type: application/sdp > Content-Length: 239 > > v=0 > o=- 1014372762 1014372762 IN IP4 192.168.13.121 > s=Asterisk > c=IN IP4 18.18.19.123 > t=0 0 > m=audio 11614 RTP/AVP 0 101 > a=rtpma...

...0@192.168.1.30;user=phone> CSeq: 1 INVITE Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37 Contact: <sip:2004@192.168.1.37:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1124335166 1124335166 IN IP4 192.168.1.37 s=Polycom IP Phone c=IN IP4 192.168.1.37 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpma...

...XXXX@192.168.1.107:5060 SIP/2.0 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE From: "XXXX"<sip:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone> Via: SIP/2.0/UDP 147.135.12.128:5060 Contact: sip:XXXX@147.135.12.128:5060 Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: <sip:XXXX@147.135.12.128>;screen=yes;party=calling;privacy=off Content-Length: 273 Content-Type: application/sdp v=0 o=2475101431 10 10 IN IP4 147.135.12.247 s=- c=IN IP4 147.135.12.250 t=0 0 m=audio 18092 RTP/AVP 0...

...;sip:7444 at 172.17.0.17> Contact: <sip:6110 at 172.17.9.1:55388;ob> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS CSeq: 24152 INVITE Route: <sip:172.17.0.17;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_d2vzw-16/r1916 Content-Type: application/sdp Content-Length: 354 v=0 o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1 t=0 0 m=audio 4006 RTP/AVP 96 3 0 8 101 c=IN IP4 172.17.9.1 a=rtcp:4007 IN IP4...

...ference being the supported: norefersub The REFER Accepted response is received by Asterisk. However, Cisco doesn't send the NOTIFY messages with 100 Trying followed by 404 Not Found. >From what we've been able to determine, this is a direct result of 200 OK packet including Supported: 100rel, timer, replaces, norefersub Specifically, the norefersub. Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190325/39eeb1e4/attachment.html>