search for: prack

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 100rel f...

...eive a call from a VoIP provider to my Asterisk which is behind a router, with 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 3...

...; <sip:0xxxxxxxx9 at sipgate.de>;tag=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...

...;tag=1c1682209279 To: <sip:3036284311@voip.livewirenet.com;user=phone> Call-ID: 1494991476221200001530@66.185.98.152 CSeq: 11 REGISTER Contact: <sip:3036284311@66.185.98.152;user=phone>;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via...

...lt;sip:777@192.168.0.13;user=phone> Contact: <sip:200@192.168.0.3;user=phone> Supported: replaces, timer Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45925 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 258 v=0 o=200 8000 8000 IN IP4 192.168.0.3 s=SIP Call c=IN IP4 192.168.0.3 t=0 0 m=audio 5004 RTP/AVP 18 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 13...

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

...m: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2 To: <sip:2000@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=s...

...0 ---> SIP/2.0 200 OK From: "asterisk"<sip:asterisk@161.49.142.250>;tag=as2cc96e52 To: <sip:192.168.45.129>;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...

...: "404" <sip:404 at 192.168.1.8>;tag=25AB8538-7BACFE71 To: <sip:10 at 192.168.1.8;user=phone> CSeq: 1 INVITE Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89 Contact: <sip:404 at 192.168.1.89> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m...

...t 192.168.0.1>;tag=as0cd1aab0 To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1 CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0 To: <si...

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

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...

...o: <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. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 10000 10000 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10....

...lt;sip:2944029@ipt.oneeighty.com>;tag=2A2425B5-B64A4132. To: <sip:2944029@ipt.oneeighty.com>. CSeq: 1 REGISTER. Call-ID: 56150889-214b0f7f-e02e6d9c@216.187.128.72. Contact: <sip:2944029@216.187.128.72>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER". User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067. Max-Forwards: 70. Expires: 3600. Content-Length: 0. . # U 216.187.140.233:5060 -> 216.187.128.72:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46. From: "Sandy Sauvage...

...;tag=as4786d518 To: <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: "Paul Ryan" <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;party=called;screen=yes;privacy=off* Contact: <sip:1098 at 192.168.62.12:5060> <sip:1098 at 192.168.62.12:5060> Content-...

...Kjxcxrrks Max-Forwards: 70 To: <sip:8500 at 10.1.0.10> From: "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=rtpm...

...uot;522" <sip:522 at 192.168.1.150>;tag=87113650-18E1B969 To: <sip:10 at 192.168.1.150;user=phone> CSeq: 1 INVITE Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99 Contact: <sip:522 at 192.168.1.99> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1207070053 1207070053 IN IP4 192.168.1.99 s=Polycom IP Phone c=IN IP4 192.168.1.99 t=0 0 m=...

...P_ADDRESS;user=phone> Call-ID: 781728720312000192946 at 192.168.105.199 CSeq: 1 INVITE Contact: <sips:201 at 192.168.105.199:5051;user=phone;transport=tls> Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003 Content-Type: application/sdp Content-Disposition: session Content-Length: 528 v=0 o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199 s=Phone-Call c=IN IP4 192.168.105.199 t=0 0 m=audio 6000 RTP/SA...

...en-1000a@hostname.company.domain>;tag=D4964260-95FB99E3 To: <sip:9990@hostname.company.domain;user=phone> CSeq: 1 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: <sip:eden-1000a@10.253.4.50> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audi...

...: <sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID: 02075d60f895e8264904b3133107aa47 at 172.16.0. 6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 1.3.4.9..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 306....v=0..o=system 8003 8000 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 172.16.0.6.. t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap :3...