search for: 5004

...RSIP during a call. The phone seems to send these MGCP RSIPs (perhaps on a schedule, looks like every 120 seconds) even when there is no call taking place. Is my diagnosis correct? What can I do about it? ...Normal RTP, then one final RTP packet and the following... 1 0.000000 192.168.0.5 5004 192.168.1.137 5004 UDP Source port: 5004 Destination port: 5004 2 0.007970 192.168.1.137 2427 192.168.0.5 2727 MGCP RSIP 14831 SHOR_001049007E83@[192.168.1.137] MGCP 1.0 3 0.008096 192.168.0.5 2727 192.168.1.137 2427 MGCP 200 14831 OK 4 0.008150 192.168.0.5 2727 192.168.1.137 2427 M...

...what > UID/GID this is? On a hunch, I tried a getfacl on a directory and triggered the error that way. Here is the result of the getfacl: # getfacl . # file: . # owner: root # group: EVOLVETSI\134domain\040users user::rwx user:root:rwx user:EVOLVETSI\134domain\040admins:rwx group::--- group:5004:r-x group:EVOLVETSI\134domain\040admins:rwx group:EVOLVETSI\134domain\040users:--- mask::rwx other::--- default:user::rwx default:user:root:rwx default:user:EVOLVETSI\134domain\040admins:rwx default:group::--- default:group:EVOLVETSI\134domain\040admins:rwx default:group:EVOLVETSI\134domain\040user...

...from the BT-101, upon the remote side answering, the call is established but no audio is passed in either direction. I have tcpdump'd this session and found this: (192.168.30.1 is * - 192.168.30.32 is BT-101) 22:41:47.899462 IP 192.168.30.32 > 192.168.30.1: icmp 36: 192.168.30.32 udp port 5004 unreachable 22:41:47.899553 IP 192.168.30.32 > 192.168.30.1: icmp 36: 192.168.30.32 udp port 5004 unreachable 22:41:47.930185 IP 192.168.30.31.8000 > 192.168.30.1.10020: UDP, length 172 22:41:47.930260 IP 192.168.30.1.10082 > 192.168.30.32.5004: UDP, length 172 22:41:47.930637 IP 192.168.3...

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

CentOS Errata and Security Advisory 2020:5004 Low Upstream details at : https://access.redhat.com/errata/RHSA-2020:5004 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 898a07f35eaa8781f7f96ee6666297c5864472115945d11efcee3bfabc801e33 resource-agents-4.1.1-61.el7_9.4.x...

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a NAT, However there are others IP Phones working fine behind NAT (from other locations, outside my networ...

...SIP # #----------------------------# function SIP_WAN { $IPT -A INPUT -p udp -i $WANIFACE --dport 5060 -j ACCEPT $IPT -A INPUT -p udp -i $WANIFACE --dport 5004 -j ACCEPT } function SIP_PORT_FORWARDING { $IPT -A PREROUTING -t nat -i $WANIFACE -p udp --dport 5060 -j DNAT --to $SIP:5060 $IPT -A FORWARD -i $WANIFACE -p udp --dport 5060 -j ACCEP...

...t=sip-extensions type=friend username=5002 callerid=5002 <5002> host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 mailbox=5002 [5003] qualify=yes context=sip-extensions type=friend username=5003 callerid=5002 <5002> host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 mailbox=5003 [5004] qualify=yes context=sip-extensions type=friend username=5004 callerid=5004 <5004> host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 mailbox=5004

...y. I am using samba 3.0.23 with Winbind and LDAP idmap backend. This server is still in testing to replace a RH9 samba server. Below I have listed the ID of three users. One is Domain Admin the others are normal users. The logs show the users initially logging in with Domain Admins rights! (GID 5004) I tried creating another group called DADMIN and changing the ownership to that and had the same result! It user would connects initially as group DADMIN. id w11350 uid=5213(w11350) gid=5004(Domain Admins) groups=5004(Domain Admins),5000(Domain Users), 5117(BUILTIN\administrators),5118(B...

...3 The data for partition 2 is: <UNUSED> The data for partition 3 is: <UNUSED> The data for partition 4 is: <UNUSED> # disklabel -r ad4s1 # /dev/ad4s1c: type: ESDI disk: ad4s1 label: flags: bytes/sector: 512 sectors/track: 63 tracks/cylinder: 255 sectors/cylinder: 16065 cylinders: 5004 sectors/unit: 80405262 rpm: 3600 interleave: 1 trackskew: 0 cylinderskew: 0 headswitch: 0 # milliseconds track-to-track seek: 0 # milliseconds drivedata: 0 8 partitions: # size offset fstype [fsize bsize bps/cpg] a: 204800 0 4.2BSD 1024 8192 16 # (...

I'm trying to set up basic monitoring for a specific extension (5004) to record all outgoing and incoming calls and save them as WAV files. I've set this in the extensions.conf file: exten => 5004,1,Answer exten => 5004,2,Wait,1 exten => 5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA LLERIDNUM}) exten => 5004,4,Monitor...

...following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-sign...

...2.168.0.3 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 headers, 13 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407...

[Posted here becasue your mail server is rejecting my direct reply to you.] Hi Martin, AFAIK SIP can run on both UDP and TCP but I have only seen it used over UDP.. :) To setup the GS phones you need to open up the following ports (If its still set at the defaults)... UDP/5060 UDP/5004 UDP/5005 UDP/5006 UDP/5007 I have not tested the GS phone through a firewall yet but this config should work.. If is doesn't try changing the ports 5004-5007 to TCP.. Hope that helps.. > Hello. > > Sorry to contact you directly, I'm on the asterisk list and asked about sip &...

...kware), half of them use a 32 bits system, the other half a 64 one. Here is a summary of the results: Bash script: Avg Min Max 32 bits 5400 4192 9024 64 bits 12900 10000 16528 Python script: Avg Min Max 32 bits 8500 5004 11132 64 bits 32800 30000 36336 R: Avg Min Max 32 bits 26900 21000 33452 64 bits 100200 93008 97496 (as a side remark, the difference between 32 and 64 is surprisingly big to me...). Then we ran the same things on our CentOS cluster, gett...

..., CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Accept: application/sdp Accept-Encoding: identity Accept-Language: en Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 212 v=0 o=10916 86824506 47524594 IN IP4 219.xx.xx.9 s=SIP CALL c=IN IP4 219.xx.xx.9 t=0 0 m=audio 5004 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 14 headers, 10 lines Destroying call '3676dea078e07a6f1e6b17084473bdea@60.xx.xx.164' Sip read: INVITE sip:192.168.1.2@60.xx.xx.164 SIP/2.0 Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4b...

...rting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning: WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 331 of extensions.conf Question: Do I need to worry about this warning? I'm a little leery of just using 366 in t...

Greetings! I'm running domain member fileservers on Samba 4.3.6.. During my backup window (backups are performed via Bacula, running on the fileserver) I tend to receive spurts of the following log message: Apr 10 16:55:18 smbfs1 winbindd[2376]: [2016/04/10 16:55:18.111192, 0] ../source3/winbindd/winbindd_group.c:45(fill_grent) Apr 10 16:55:18 smbfs1 winbindd[2376]: Failed to find domain

...00 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: 302 v=0 o=1310 8000 8000 IN IP4 pickup.phone.ip.addr s=SIP Call c=IN IP4 pickup.phone.ip.addr t=0 0 m=audio 5004 RTP/AVP 0 8 3 4 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 13 headers, 15 lines Using latest request as basis request Sending to pickup.phone.ip.addr : 5060...

...E,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp> Supported: replaces User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (16 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060...