search for: vvvr

Hi guys, I'm having a problem rsyncing between two directories on local disk. I wish to sync the local directory /local/CA/eTrustDirectory/dxserver/ldif to the other local directory /local/CA/archive. 'rsync -vvvr --progress /local/CA/eTrustDirectory/dxserver/ldif /local/CA/archive' pauses at: make_file(4,ldif/MELLDRBIB11-ldsau004wm1/ldbau004v1_20060330.0204.raw) make_file(4,ldif/MELLDRBIB11-ldsau004wm1/ldbau004v1_20060331.0204.raw) 782 files to consider For at least 50 minutes (I cancel it at this poin...

...d argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Another configuration error with the software (or being generous, perhaps it's because it's the wrong software for the kit). [root@carol asterisk]# asterisk -vvvr ERROR[16384]: File asterisk.c, Line 1323 (main): Unable to connect to remote asterisk Hmm...that wasn't happening before I did the build using the supplied disc. Well, if anyone has the correct install disc, can they copy and zip/tgz it for me. Otherwise, anyone have the X100P & TDM40...

d# rsync -vcrlpogtz . rsync@domain.ltd:/export/home/rsync/ rsync@domain.ltd's password: sh: rsync: rsync: connection unexpectedly closed (0 bytes read so far) rsync error: error in rsync protocol data stream (code 12) at io.c(189) d# Can someone tell me what am I doing wrong? Thanks

...edia path and breaking my ability to do the intra-office transfer. According to what I find in teh mailing list archives, putting a T/t as an option to dial() will halt a possible transfer and keep the system in the media path. However, that doesn't seem to be the case. I ran "asterisk -vvvr" to watch the call being processed and I can see the DIAL(<channel>||T) be called and shortly thereafter it gives the "Ready to transfer" and then indicates the hangup while the other two * systems are handling the channel. So what I see happening is not what the docs and arch...

Hi, On the release of asterisk I was using before this one, I used to issue a "set verbose 100" command and I would see all the sip registrations taking place. Now that doesn't seem to work. Could someone clarify what value I should use with the "set verbose" command in order to see sip registrations. Thanks -------------- next part -------------- An HTML attachment was

Hello, After testing some different phones, codecs and combinations of them I noticed that some of my GSM applications didn't work anymore. So after finding out what it was (no codec support) I was thinking, why * couldn't give my that error directly (-d -vvvr did not give any feedback). One thing for sure, it didn't do any codec translation or gave an error about that. An error if -> an application <- can't agree upon codec would be nice, is there allready a way to see this 'newbie' SIP errors? Stefan

Hi all, for an unknown reason, I find my asterisk server down every morning as a result of failing to restart during the night because of a segmentation fault. The error message is as follows: Waiting for inactivity to perform restart... Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). -- Remote UNIX connection Waiting for

...============ Just to check, I added the actual filename (.so): ================ > cat modules.conf noload => codec_speex.c noload => codec_speex.so ip04*CLI> reload ip04*CLI> show modules codec_speex.so > /etc/init.d/asterisk stop > /etc/init.d/asterisk start > asterisk -vvvr ip04*CLI> show modules codec_speex.so ================ Does someone know why Asterisk still loads modules even with the above lines in modules.conf? Thank you.

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

Does someone know coincidentally the cause for the error message specified in the Subject? The following scenario: Snom 360 behind one rout (wiederrum on a DSL line with static IP address hangs). The Snom has a private IP, routs accomplishes NAT. STUN and ICE are activated, as SIP haven 5060/udp are firmly used. Detailed packages passed on on haven 5060/udp of rout to the Snom. The telephone

Hello, I've recently installed Asterisk@home, i'm following step by step the "new user guide" but I cannot get my X-Lite SIP phone see my asterisk@home proxy... I've installed in a viertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia -------------- next part

...yback looks malformed based upon the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Playback) Use this instead.... exten => s,2,Playback(demo-thanks) ;for playing a file Stop * from the CLI and then restart using the following... asterisk <press enter> astersik -vvvr <press enter> Now issue the ZAP SHOW CHANNELS command again. If your ztcfg -vv gave you channels, they should be working. The order you compiled everything in is important too... http://www.automated.it/guidetoasterisk.htm#_Toc49248761 -----Original Message----- From: khoonking [mailto...

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=...

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP