search for: gruu

...cord-Route: <sip:abpusa.com:5060;maddr=209.189.239.106> From: <sip:9723048722@abpusa.com;user=phone>;tag=wr4771pry1 To: <sip:9723048720@abpusa.com>;tag=gh50agmbxb Call-ID: 3c26700d3618-u5uzu5ev2xn6@4-12-220-193 CSeq: 9 NOTIFY Max-Forwards: 69 Contact: <sip:9723048722@abpusa.com;gruu=dekqyehe> Event: dialog Content-Type: application/dialog-info+xml Content-Length: 556 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="8" state="full" entity="sip:9723048722@abpusa.com;user=phone"&g...

...dio;mobil= ity=3D"fixed";duplex=3D"full";description=3D"snom320";actor=3D"principal";e= vents=3D"dialog";methods=3D"INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBS= CRIBE,PRACK,MESSAGE,INFO"=0D=0D User-Agent: snom320/7.1.8=0D=0D Supported: gruu=0D=0D Allow-Events: dialog=0D=0D X-Real-IP: 192.168.0.4=0D=0D Expires: 3600=0D=0D Content-Length: 0=0D=0D =0D=0D =0D <------------->=0D =00--- (14 headers 0 lines) ---=0D =00Using latest REGISTER request as basis request=0D =00Sending to 192.168.0.4 : 2048 (NAT)=0D =00=0D=1B[Kjdc*CLI> =0D=...

...quot;;duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.31 Allow-Events: dialog X-Real-IP: 192.168.101.102 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.101.102:2061 (no NAT) <--- Transmitting (NAT) to 192.168.101.102:2061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;received=192.168.101.10...

...: "Mike's Peoria Snom" <sip:MikePeoriaSnom___1@198.88.216.85> Call-ID: 3c4c743d4841-8oaq4jsghi78@192-168-1-2 CSeq: 106321 REGISTER Max-Forwards: 70 Contact: <sip:MikePeoriaSnom___1@192.168.1.2:5060;line=d313mbjt>;q=1.0 User-Agent: snom200-3.56m P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.2 WWW-Contact: <http://192.168.1.2:80> WWW-Contact: <https://192.168.1.2:443> Expires: 60 Content-Length: 0 17 headers, 0 lines Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Jan 13 18:48:41 WARNING[7570]: res_config...

...TER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: <sip:5441 at 172.16.99.161:5060;transport=udp>;+sip.instance="<urn:uuid:000 00000-0000-1000-8000-00085D13D80D>";expires=3600 Supported: gruu User-Agent: Aastra 6739i/3.0.1.38 Content-Length: 0 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.100.250.10:5060;branch=z9hG4bK46044b1bcfd826368;received=172.16.99.16 1 From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb To: <sip:5441 at 10.100.250.10:5060> Call-ID: 22fd19f44649a4...

...BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "SIP Phone - Ext. 202" <sip:202 at 192.168.50.5:5060 ;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D25B72F>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172...

...313e82f4af9423bab056113e5e05713 To: <sip:3 at myhost.org> Contact: <sip:03794281 at 192.168.1.2:51861> Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:99 spe...

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

....168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129> Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 CSeq: 45683 REGISTER Max-Forwards: 70 Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5>;q=1.0 User-Agent: snom190-3.56i P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.102.70 WWW-Contact: <http://192.168.102.70:80> WWW-Contact: <https://192.168.102.70:443> Expires: 60 Content-Length: 0 ------------------------------------------------------------------------ Received from udp:192.168.0.129:5060 at 14/12/2004 18...

...ot;;duplex="full";description="snom870";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom870/8.7.3.25.5 Allow-Events: dialog X-Real-IP: 192.168.6.112 Supported: path, gruu Expires: 3600 Content-Length: 0 The SNOM-870 is sending registration via UDP and not by TLS. Is that how things are supposed to work? If only TLS is enabled in Asterisk for that peer then evidently the peer cannot register. But is registration supposed to be done via TLS? If so then how does...

...;<div>From: "G" <sip:8000@177.64.122.237>;tag=ue84kn6rku</div><div>Call-ID: u5hkiispkvn9g841oede</div><div>CSeq: 9338 BYE</div><div>Reason: SIP ;cause=488; text="Not Acceptable Here"</div><div>Supported: path, outbound, gruu</div><div>User-Agent: JsSIP 0.3.7</div><div>Content-Length: 0</div><div><br></div><d iv>Some one can help me with this problem?</div><div><br></div><div>Thanks </div><div><br></div><div&g...

...t;sip:snom_01@192.168.0.129> >>Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 >>CSeq: 45683 REGISTER >>Max-Forwards: 70 >>Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5>;q=1.0 >>User-Agent: snom190-3.56i >>P-NAT-Refresh: 15 >>Supported: gruu >>Allow-Events: dialog >>X-Real-IP: 192.168.102.70 >>WWW-Contact: <http://192.168.102.70:80> >>WWW-Contact: <https://192.168.102.70:443> >>Expires: 60 >>Content-Length: 0 >> >>---------------------------------------------------------------...

...t;sip:11 at woody.ch> Call-ID: 6lrsku1p at snom CSeq: 933701145 SUBSCRIBE Max-Forwards: 70 Contact: <sip:11@[2001:4060:dead:d1d0:204:13ff:fe30:228d]:2799;transport=udp;line=u5go22>;reg-id=1;+sip.instance="<urn:uuid:572d1aa1-bfd5-4b8a-ab1e-f9095df386e5>" Supported: outbound, gruu Event: message-summary Accept: application/simple-message-summary User-Agent: snom-m9/9.6.13-a Authorization: Digest realm="asterisk",*** remaining line removed for this email *** Expires: 60 Content-Length: 0 2017/12/02 11:58:00 [SIP-Reg:5]: MWI subscription on identity 1 failed. Retry...

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

...TE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" <sip:85014 at 192.168.96.141:5060 ;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2B85C3>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 6731i/2.6.0.1007 Content-Type: application/sdp Content-Length: 698 v=0 o=MxSIP 0 0 IN IP4 192.168.96.141 s=SIP Call c=IN IP4 192.168.96.141 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101 a=rtpmap:0 PCMU/8000 a=...

...quot;;duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/8.4.18 Allow-Events: dialog X-Real-IP: 192.168.114.200 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] --- (14 headers 0 lines) --- [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] Sending to 192.168.114.200 : 2048 (no NAT) [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c...

...nd RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.211.0.42:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 101-test SIP Options : 100rel gruu path replaces replace timer Codecs : 0x106 (gsm|ulaw|g729) Codec Order : (g729:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (34 ms) Useragent : Aastra 55i/2.6.0.1008 Reg. Contact : sip:101-test at 10.211.0.42:5060;transport=udp Qualify Freq : 6...