search for: c0a80101

...CoS mark 4 Here Rebooting/Power up Thomson 2030S on 192.168.11.72 <--- SIP read from UDP:192.168.11.72:5060 ---> REGISTER sip:192.168.11.251;user=phone SIP/2.0 v: SIP/2.0/UDP 192.168.11.72:5060;branch=z9hG4bK8030819752698542643-155044 f: <sip:41 at 192.168.11.251:5060;user=phone>;tag=c0a80101-25da4 t: <sip:41 at 192.168.11.251:5060;user=phone> i: 5789b-c0a80101-5-4 at 192.168.11.72 CSeq: 1 REGISTER Route: <sip:192.168.11.251:5060;lr> Max-Forwards: 70 Expires: 60 m: <sip:41 at 192.168.11.72:5060;user=phone> User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-5A-AA Allow-E...

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target ch...

<--- SIP read from 192.168.32.245:5060 ---> SIP/2.0 481 CallLeg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: "asterisk"<sip:asterisk at 192.168.32.16>;tag=as2ff08179 To: <sip:5386 at 192.168.32.245:5060;user=phone>;tag=c0a80101-2ce1bc03 Call-ID: 2fa28b4-c0a80101-d-9acc at 192.168.32.245 CSeq: 143 NOTIFY Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.32.245:5060: NOTIFY sip:5386 at 192.168.32.245:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: "asterisk&...

...lso as a check for a mailbox's existence (though corresponding mailbox is defined in voicemail.conf). SUBSCRIBE sip:*foo*@192.168.100.254:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.193:5060 ;branch=z9hG4bK8031919793092643643-24593 From: <sip:7533 at 192.168.100.254:5060>;tag=c0a80101-600e To: <sip:7533 at 192.168.100.254:5060> Call-ID: c001-c0a80101-d-2 at 192.168.100.193 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Event: message-summary Accept: application/simple-message-summary Expires: 3600 Contact: <sip:7533 at 192.168.100.193:5060;user=phone> Allow-Events: refe...

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello: > What do you mean now? If I can use the full available band or if I can > download exactly 50Mbs? > The answer to the first question is: YES! That's why I use a traffic > shaper... ;) > The answer to the second question is: NO. I made a speedtest right now > and I get only ~18Mbps download. And some other information, too.

...many packets headed out.  Our tool takes a shot at it: jeff at jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I testPhone.pcap input: testPhone.pcap 2020/06/16 10:47:16.047401 INVITE 192.168.200.10:25572 (Luca) -> 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,10000) 2020/06/16 10:47:16.112866 DUPINVITE 192.168.200.10:25572 (Luca) -> 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,10000) 2020/06/16 10:48:43.690647 BYE 192.168.200.1:25572(sip:035014649215) -> 192.168.200.10:25572(Luca)     Session: 81b175...

...TE 0049177xxxxxxx 217.0.27.53 00493501xxxxxxx 5647efe41d746b4 (alaw) No Tx: INVITE pbxluca 2 active SIP dialogs > And "sip show channel <Call-ID>" for more info. Call from normal phone: bpi*CLI> sip show channel 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 * SIP Call Curr. trans. direction: Incoming Call-ID: 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 Owner channel ID: SIP/0049351xxxxxxx-000000a7 Our Codec Capability: (alaw|ulaw|ilbc|g729|g723|gsm) Non-Codec Capability (DTMF):...

...on: /----------------------------------------------------------------------- ----------------------- | SIP/2.0 200 OK | Via: SIP/2.0/UDP 212.147.65.204:5060 | From: "252"<sip:252@192.168.1.35:5060> | To: <sip:252@192.168.1.35:5060>;tag=c0a80101-23004a | Call-ID: 400842155@212.147.65.204 | CSeq: 4660 OPTIONS | Contact: <sip:252@192.168.1.35:5060;user=phone> | Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGIST ER,INFO | Supported: timer, replaces |...

...53 00493501xxxxxxx 5647efe41d746b4 (alaw) No > Tx: INVITE pbxluca > 2 active SIP dialogs > > >> And "sip show channel <Call-ID>" for more info. > > Call from normal phone: > > bpi*CLI> sip show channel 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 > > * SIP Call > > > Curr. trans. direction: Incoming > > > Call-ID: 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 > > > Owner channel ID: SIP/0049351xxxxxxx-000000a7 > Our Codec Capability: (alaw|ulaw...

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

...) replies with : SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport From: "9121 Guest1"<sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254> >;tag=as237a9159 To: <sip:9121 at 192.168.100.198:5060;user=phone>;tag=c0a80101-a611e Call-ID: 6bddeb200c2aee553856dab4098c6f8e at 192.168.100.254 CSeq: 102 INVITE Content-Length: 0 Is it normal to reply this way ? I tried with another SIP phone (a Siemens Gigaset) and it accepted the INVITE (and started to ring). regards -------------- next part -------------- An HTML at...

...n 7791 is calling extension 7792, I can see INVITE messages coming in and out Asterisk. I can also see a NOTIFY message advertising this call to subscriber 7793, for instance. Here is an extract of this message : NOTIFY sip:7793 at 192.168.101.102:5060;user=phone SIP/2.0 <snip> Call-ID: 7019-c0a80101-d-3 at 192.168.101.102 <snip> Content-Length: 212

...read whatever messages might be showing up. All I can see is that the screen is filled about 2/3 way down before the reset. Here are the relevant portions of the tftpd log: tftpd[593]: providence: read request for //pxelinux.0: success tftpd[595]: providence: read request for //pxelinux.cfg/C0A80101: success Note that it's not trying to load the kernel image at all. It did occur to me that this might be one of the tftpd-related problems mentioned in the docs, but unfortunately, the docs are extremely thin on what non-Linux tftpds _do_ work correctly (presumably because you don't know...

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123