search for: byes

I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?

I'm getting strange timeouts occasionally on Dovecot 1.0.rc15 on Debian 4.0, and haven't been able to figure out the cause. $ ./src/imaptest user=test%d at imaptest.com host=localhost pass=foo mbox=~/dovecot-crlf - select=0 Logi Logo 100% 100% 2 4 7/ 10 0 0 10/ 10 0 0 10/ 10 0 0 10/ 10 0 0 10/ 10 0 0 10/ 10 0 0 10/ 10 0 0

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

Dear All, I have the following setup running perfectly OK for a long time CentOS release 5 (Final) sendmail-8.13.8-2.el5 MailScanner 4.76.25 bind-9.3.4-6.0.3.P1.el5_2 now i jus setup a centos box running BackupPC for backing up my my above mail server using ssh as per the instructions in backup pc site i had to enable sshd so i did it and everthing works perfect and backup works great as per my

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

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 (5060) ------------------...

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 it from scratch. This

I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip 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

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

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

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

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

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

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, > >

OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xxxxxx Password: 1000xxxxxx Server: brxxxx.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

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+ /

Ok. I've noticed a thing: when you ring a sip phone, and 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