search for: bye

...6478517573 2752611-195 00101/00001 ulaw No Rx: ACK 136.59.30.19 8787041796 76775e35788 00102/00000 ulaw No Tx: ACK 9.9.95.13 9057047798 2752419-199 00101/00001 ulaw No Rx: ACK 195.7.123.234 +011503733 25afde8070b 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011503733 71688696061 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011503733 1700ab8b2ae 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011578435 0ecb33f75bb 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011962642 71eac20715c 00102/00002 unkn...

...r 16 secs in command: 2 LOGIN "test1 at imaptest.stage.dealerflow.com " "test" - 12 stalled for 16 secs in command: 2 LOGIN "test1 at imaptest.stage.dealerflow.com " "test" Eventually, I get: Error: test18 at imaptest.stage.dealerflow.com[6]: Unexpected BYE: * BYE Disconnected for inactivity. Error: test43 at imaptest.stage.dealerflow.com[10]: Unexpected BYE: * BYE Disconnected for inactivity. Error: test19 at imaptest.stage.dealerflow.com[9]: Unexpected BYE: * BYE Disconnected for inactivity. Error: test40 at imaptest.stage.dealerflow.com[8]: U...

Hi, from log: Jul? 4 12:10:19 localhost dovecot: master: Warning: service(imap): process_limit (2) reached, client connections are being dropped Jul? 4 12:10:19 localhost dovecot: log(1078): Error: Received master input for invalid service_fd 22: 22 9009 BYE Jul? 4 12:10:58 localhost dovecot: master: Warning: service(imap): process_limit (2) reached, client connections are being dropped Jul? 4 12:10:58 localhost dovecot: log(1078): Error: Received master input for invalid service_fd 22: 22 9012 BYE Jul? 4 12:11:58 localhost dovecot: log(1078): Error:...

...og/secure i see the followin very often --------------------------- Jun 19 16:26:06 kmdns1 sshd[11073]: Invalid user jeka from 87.118.122.78 Jun 19 16:26:06 kmdns1 sshd[11074]: input_userauth_request: invalid user jeka Jun 19 16:26:06 kmdns1 sshd[11074]: Received disconnect from 87.118.122.78: 11: Bye Bye Jun 19 16:26:07 kmdns1 sshd[11075]: Invalid user stat from 87.118.122.78 Jun 19 16:26:07 kmdns1 sshd[11076]: input_userauth_request: invalid user stat Jun 19 16:26:08 kmdns1 sshd[11076]: Received disconnect from 87.118.122.78: 11: Bye Bye Jun 19 16:26:09 kmdns1 sshd[11077]: Invalid user nikonew...

Hi, My dovecot process seam blocked on dovecot/imap [blocking on log write], only restart fix it. How solve that's? Cheers, -- alpha_one_x86/BRULE Herman <alpha_one_x86 at first-world.info> Main developer of Supercopier/Ultracopier/CatchChallenger, Esourcing and server management IT, OS, technologies, research & development, security and business department -------------- next

Hi there noticed a strange thing in asterisk 1.6.2x 1.6.1x after one of the clients sends bye asterisk first sends invite to other side then after 200 ok it sends bye I am not sure but that could be some missconfiguration issue or a bug? so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK fr...

Hi all, I have a lot of call in wich I found that my Asterisk doesn't answer the BYE 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 (5...

I'm a bit confused about how * constructs the RURI when it generates a BYE. For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs...

...ip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7f...

I see many entries in /var/log/secure similar to these: . . . /var/log/secure.1:Dec 31 08:00:55 gway01 sshd[7220]: Received disconnect from 93.89.144.31: 11: Bye Bye /var/log/secure.1:Dec 31 08:00:58 gway01 sshd[7221]: Failed password for root from 93.89.144.31 port 60100 ssh2 /var/log/secure.1:Dec 31 08:00:58 gway01 sshd[7222]: Received disconnect from 93.89.144.31: 11: Bye Bye /var/log/secure.1:Dec 31 08:01:02 gway01 sshd[7223]: Failed password for root f...

...ACK / 102 ACK 14. TxReqRel INVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRel INVITE / 104 INVITE 22. Rx BYE / 302 BYE 23. TxResp SIP/2.0 / 302 BYE 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy Why is asterisk allowing an invite after receiving a bye on a particular session/channel? From what I've read.. a bye should be the termination of the session/channel and therefore it s...

Hi all, I've read several discussions about asterisk adding 'received' parameter to the top Via header. In our case asterisk (release 11.4) gets BYE from sip proxy (with BYE top via header containing proxy ip address and port) but added 'received' parameter contains ip address from a 2nd Via (or from "From') and OK gets lost. I'm just trying to adjust sip configuration that used to work for simple call scenarios (in 1.4, f...

Hi gurus, In need of a little help here. I?m trying to do the Asterisk media release by using canreinvite=yes. But I found weird behaviour when comes to BYE. Below are my current setup: Client A is registered to Opensips Client B is registered to Asterisk A ? Opensips ? Asterisk ? B On hangup below are the SIP flow which I?ve notice from the Asterisk server itself: 1. Opensips forward the BYE to Asterisk 2. Asterisk response with 200 OK 3. Asterisk...

...r=phone>;tag=1c1682209279 To: <sip:3036284311@voip.livewirenet.com;user=phone> Call-ID: 1494991476221200001530@66.185.98.152 CSeq: 11 REGISTER Contact: <sip:3036284311@66.185.98.152;user=phone>;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 1...

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote: > ** > Hi Nick, > > The BYE is not properly formed and rejected by script - in the 200 OK of > the INVITE, you can see that your opensips is doing Record-Routing, but the > BYE does not contain the corresponding Route hdr, so SIP routing is > impossible. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS...

...1CA65AC-9C8 To: <sip:84104214@203.88.192.42> Date: Thu, 29 Sep 2005 20:14:40 GMT Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF@211.147.240.237 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2153363387-811340250-2169109749-53752559 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: <sip:0017911@211.147.240.237>;party=calling;screen=yes;privacy=off Timestamp: 1128024880 Contact: <sip:0017911@211.147.240.237:57786> Expires: 180 Allow-Events: telephone-event C...

...2.70;rport=5060 From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129>;tag=as1a8337d5 Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 CSeq: 45683 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:snom_01@192.168.0.129> Content-Length: 0 ------------------------------------------------------------------------ Received from udp:192.168.0.129:5060 at 14/12/2004 18:21:29:580 (513 bytes): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-w...

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+ /

...d hangup before it answer, asterisk sends a CANCEL to the phone to abort the current operation (in this case, the INVITE). and this's correct according to rfc. But now... when a sip phone A is ringed from a phone B , and that call from B is picked up by the phone C via *8 , asterisk sends 'BYE' to the phone A ( C & B are bridged ok). But according to rfc, that's wrong, since 'BYE' must be sent to release an active call . The right thing to do is to send a CANCEL to A, since we want to abort the pending INVITE. I'm right ? That's a bug in asterisk ? I've...