search for: rinstance

...DY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.1.2:34312: INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport Max-Forwards: 70 From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> >;tag=as2886cf30 To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> Contact: &lt...

hello i was wondering what is the use of "rinstance" in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case "rinstance" parameter is missing. when i dial "SIP/username" audio is fine but when i dial...

...the manager API. There is no video, either. Any idea why video doesn't work for outgoing call from asterisk? cwhuang*CLI> sip set debug peer 403 SIP Debugging Enabled for IP: 172.16.148.129:36042 Reliably Transmitting (NAT) to 172.16.148.129:36042: OPTIONS sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0 Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK0dcbc40d;rport From: "asterisk" <sip:asterisk at 172.16.148.1>;tag=as25cd8375 To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901> Contact: <sip:asterisk at 172.16.148.1> Call-ID: 18ce6...

...is IP:Port (sip uri), call is connected fine and he can hear the called user but the called user can not here the caller voice. If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. what i have noticed is that IP:Port dial is missing a parameter "rinstance" in "Contact" , "To" headers for adf. what is "rinstance" for? Also something with "Contact" header seems fishy. or RTP issue. that is Dial(SIP/adf,30,r) works fine with bothway audio, but Dial(SIP/116.18.35.235:28614,30,r) has one way audio....

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

..._config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'via_port' cannot be type 'int(11)' (need char) [Sep 2 15:38:46] ERROR[2098]: res_pjsip_registrar.c:411 register_aor_core: Unable to bind contact 'sip:2001 at 192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525' to AOR '2001' == Contact 2001/sip:2001 at 192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525 has been deleted The mysql warnings have always been there since version 13.0 and the "Unable to bind contact..." error has also been present s...

...ch it was registered. like this, Dial(SIP/XYZ at 117.58.x.x:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). after taking debug trace of both methods what I found was that a SIP HEADER parameter "rinstance" was missing in "to" and "INVITE" header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0 To: <sip:XYZ at xxxxxxxxxxxx:28614;rinstance=02...

...6 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379 send_unsolicited_mwi_notify_to_contact: Unable to create unsolicited NOTIFY request to endpoint 1001 at sip.domain.com URI sip:1001 at 95.250.29.3:50673;rinstance=1af959e7c0059fc4 Regards El 16/05/2016 a las 02:52, George Joseph escribi?: > > > On Sun, May 15, 2016 at 12:00 PM, Annus Fictus <annusfictus at gmail.com > <mailto:annusfictus at gmail.com>> wrote: > > Hello List, > > following this thread: > >...

...UzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:85236418505@10.201.0.224>. Content-Length: 0. . # U 10.201.0.224:5060 -> 10.0.0.25:2750 OPTIONS sip:9000220002@10.0.0.25:2750;rinstance=f136277835976893 SIP/2.0. Via: SIP/2.0/UDP 10.201.0.224:5060;branch=z9hG4bK3069abdf;rport. From: "asterisk" <sip:asterisk@10.201.0.224>;tag=as77042273. To: <sip:9000220002@10.0.0.25:2750;rinstance=f136277835976893>. Contact: <sip:asterisk@10.201.0.224>. Call-ID: 5dae5713...

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

...the Asterisk console: ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound OPTIONS request to endpoint 1000 at sip.domain.com ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to create request to qualify contact sip:1000 at 95.250.29.3:53570;rinstance=d90827763e4353c0 in the aor section I'm using: qualify_frequency=30 Any hint? Regards

...t=TCP Transport: TCP Sent-by Address: 192.168.1.15 Sent-by port: 47053 Branch: z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z- RPort: rport transport=TCP Max-Forwards: 70 Contact: <sip:5678 at 192.168.1.15:47053 ;rinstance=bea6f11f37c55605;transport=TCP> Contact URI: sip:5678 at 192.168.1.15:47053 ;rinstance=bea6f11f37c55605;transport=TCP Contact URI User Part: 5678 Contact URI Host Part: 192.168.1.15 Contact URI Host Port: 47053 Contact U...

...d the messages, I got below. Please help me in solving the problem. Thanks in advance, Treesa REGISTER sip:192.168.12.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f- 1-- d87543-;rport Max-Forwards: 70 Contact: <sip:1002 at 192.168.25.116:52166;rinstance=1a12ef13351e0ee1> To: "1002"<sip:1002 at 192.168.12.160> From: "1002"<sip:1002 at 192.168.12.160>;tag=5f799517 Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SU...

...working perfectly from the ATA (linksys pap2t) but not from asterisk, because it hangs up after 10 seconds. Some LOGS: [Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113 with 192.168.20.0 [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" < sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.168...

...tration fails. Another remark : when using a Zoiper softphone, the registration goes very well : REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z- Max-Forwards: 70 Contact: <sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP> To: <sip:test3 at sip.domain.tld;transport=UDP> From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE,...

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

...-- Executing [702 at from-internal:2] NoCDR("SIP/701-0864f1b8", "") in new stack -- Executing [702 at from-internal:3] Wait("SIP/701-0864f1b8", "1") in new stack Retransmitting #1 (NAT) to 123.456.789.000:9855: OPTIONS sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK782c5851;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown at 192.168.1.4 <sip%3AUnknown at 192.168.1.4> >;tag=as43db5836 To: <sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907> Contact:...

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

...eone can help me to use this line with my asterisk. These are the traces of my Xllite REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-e322ee549824f666-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:8887776666 at 10.0.0.221:22818;rinstance=570ac597afa82c9a> To: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022> From: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022>;tag=fb1acd4f Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk. CSeq: 2 REGISTER Expires: 3600 Allow: I...

I''ve posted a lengthy description of an authorization plugin for Rails on my blog: http://www.billkatz.com/authorization It describes a proposed DSL for authorization, a pattern for use that describes conventions, and a reference implementation that lets you test the some of the ideas. I hope that some subset of Rails programmers gravitate toward a common DSL for authorization, which