search for: 5060

...E message, then the BYEs are retransmitted, but the call ends, when the Asterisk sends a BYE. Time AS.TE.RI.SK CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From: sip:1265666072 at 81.209.186.14 <sip%3A1265666072 at 81.209.186.14>To:sip:1234567890 at CA.RR.IE.R1 (5060) ------------------>71(5060)1U telephone-event) 0.09 (5060) 100 Trying-------->71(5060)1U telephone-event) (5060) <------------------71(5060)1U telephone-event) 5.18 (5060) 183 Session Progress SDP ( g729 telephone-event)1(5060)1U telephone-event) (5060) <------------------s (5060...

...e | Client | Asterisk | | | | OpenSIPS | |7.158764 | INVITE SDP (g711U g7 | |SIP From: "760xxxxxxx" <sip:760xxxxxxx at client To:<sip:317xxxxxxx at OpenSIPS | |(5060) ------------------> (5060) | | |7.159003 | | INVITE SDP (g711U g7 |SIP Request | | |(5060) ------------------> (5061) | |7.161857 | | 100 Trying| |SIP Status |...

...roblem? |Time | 209.220.119.18 | | | | 208.88.61.150 | |9687.369 | INVITE SDP (g729 telephone-eventRTPType-101) |SIP From: <sip:vmax at 209.220.119.18 To:<sip:17329071590 at 208.88.61.150 | |(5060) ------------------> (5060) | |9687.390 | 100 Trying| |SIP Status | |(5060) <------------------ (5060) | |9687.394 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status | |(5060) <------------------ (5060) | |96...

...2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This results in dropping both calls. I've attached a sanitized sip trace from the snom phone for your perusal. Thanks for any help you can offer. Brian ### START SIP TRACE ### Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 bytes): REGISTER sip:192.168.0.129 SIP/2.0 Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129> Call-ID: 3c267319f1...

...[5878 at phones:1] Dial("SIP/5253-0823eab0", "SIP/5878") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 10.7.10.1 port 19968 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.7.10.51:5060: INVITE sip:5878 at 10.7.10.51:5060 SIP/2.0 Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport Max-Forwards: 70 From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a To: <sip:5878 at 10.7.10.51:5060> Contact: <sip:5253 at 10.7.10.1:5060> Call-ID: 7903ae4900c136a4...

...s mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail -------------- next part -------------- Sip read: REGISTER sip:<asterisk ip>:5060;transport=udp SIP/2.0 Call-ID: 18883d642b2e1730629c792a35bb0fb6@172.16.1.54 CSeq: 1 REGISTER From: "3002" <sip:3002@<asterisk ip>:5060;transport=udp>;tag=24957277 To: "3002" <sip:3002@<asterisk ip>:5060;transport=udp> Via: SIP/2.0/UDP 172.16.1.54:5060;bra...

...xtensions.conf: [from-internal] ... exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r)) I tried a echo test, dialing in my case to 8500, but in spite of seeing traffic towards Internet, nothing is heard. Setting 'sip set debug', I get the following thing: <--- SIP read from 10.1.0.65:5060 ---> INVITE sip:8500 at 10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks 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: <...

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+ /

...since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060 | |(2752) ------------------> (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060 | |(2752) ------------------> (5060)...

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type = friend amaflags = billing accountcode = SIPUSER disallow=all allow=ulaw allow=alaw [1234] host = dynami...

...dio 7200 RTP/AVP 8 101 [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found I enabled debug on the IP of the dect-phone (full log attached), but it does not make me any wiser... set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port to send to set_destination: set destination to 192.168.0.27:5060 Reliably Transmitting (no NAT) to 192.168.0.27:5060: BYE sip:dect at 192.168.0.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8 Max-Forwards: 70 From: "fam-witvliet eerste verdiep&q...

Typically (or at least somewhat occasionally) after a reboot of my shorewall[-lite] machine I find that I end up with conntrack table entries for unNATted connections such as: # conntrack -L -p udp --dport 5060 -d 99.232.11.14 udp 17 59 src=10.75.22.8 dst=99.232.11.14 sport=5060 dport=5060 packets=5472 bytes=3031488 [UNREPLIED] src=99.232.11.14 dst=10.75.22.8 sport=5060 dport=5060 packets=0 bytes=0 mark=1 use=2 These are supposed to be NATted and will be so if I flush the offending entries from the...

...and accurate. Does anyone have any idea what might be wrong here? FAILED CALL: Conv.| Time | MY_HOST | 64.2.142.215.GIGe-net.vitel.net| 6 |186.532 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR To:sip:15555551212 at outbound.vitelity.net | |(5060) ------------------> (5060) | 6 |186.575 | 407 Proxy Authentication Required |SIP Status | |(5060) <------------------ (5060) | 6 |186.780 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR To:sip:15555551212 at out...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

...ried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer. Below is the SIP debug Thank for any help.... to 162.33.165.195:5060 Sip read: INVITE sip:14103445557@162.33.165.198 SIP/2.0 Record-Route: <sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176> Via: SIP/2.0/UDP 213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1 Via: SIP/2.0/UDP 213.137.65.234:5060 From: <sip:4103532264@213.137.65.234>...

...in to my A*k server, and I see "Got SIP response 405 "Method Not Allowed" back from 192.168.3.64" but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- -- Got SIP response 489 "Bad event" back from 192.168.3.10 ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines)...

...phone. This results in dropping both calls. >> I've attached a sanitized sip trace from the snom phone for your >>perusal. >> >>Thanks for any help you can offer. >> >>Brian >> >>### START SIP TRACE ### >> >>Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 bytes): >> >>REGISTER sip:192.168.0.129 SIP/2.0 >>Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport >>From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi >>To: "snom_01" <sip:snom_...

...n by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555 at 64.2.142.93> Contact:...

...IP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI> sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:...

...brief power outage last Friday, most phones went down but the PBX stood up (power generator). Anyhow, right now all of the other phones work just fine, but the Cisco's are acting up. My return time is: (numbers sanitized) (sip show peers) 1133/1133 10.10.2.85 D 5060 OK (69 ms) 2217/2217 10.10.2.81 D 5060 OK (15 ms) 2222/2222 10.10.2.82 D 5060 OK (16 ms) 1137/1137 10.10.2.203 D 5060 OK (202 ms) 3329/3329 10.10.2.51 D 506...