search for: 4444

...02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, thelma at sys-concept.com wrote: > > <snip> > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn-4444 >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-4444" in sip.conf? > > It found pstn-4444 using 10.10.0.8:5060 - if the request always comes from the same IP add...

...t sys-concept.com wrote: >>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >>>> >>>> IN audocodes setting I have: >>>> "EndPoint Phone Number" >>>> >>>> Channel: 3 phone number: pstn-4444 >>>> Channel: 4 phone number: pstn-9998 >>>> >>>> When I am calling " pstn-4444" the port number "Channel:3" lights up but >>>> asterisk is showing that the call is coming on "pstn-9998" >>>> >>>&g...

...> On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998 >> >> When I am calling " pstn-4444" the port number "Channel:3" lights up but >> asterisk is showing that the call is coming on "pstn-9998" >> >> -- Executing ..... Answer("SIP/pstn-9998...

...o know more than 1 single extension. I have it setup to the point that I have created the voicemail box and the incoming call gets directed to it but I then get the error: Aug 10 10:40:48 DEBUG[1906]: Exiting with DIALSTATUS=NOANSWER. Aug 10 10:40:48 VERBOSE[1906]: -- Executing Goto("IAX2/4444@4444/2", "custom-1000|s|2") in new stack Aug 10 10:40:48 VERBOSE[1906]: -- Goto (custom-1000,s,2) Aug 10 10:40:48 VERBOSE[1906]: -- Executing Goto("IAX2/4444@4444/2", "exten-vm|1000@default|1000") in new stack Aug 10 10:40:48 VERBOSE[1906]: -- Goto (ex...

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3 phone number: pstn-4444 Channel: 4 phone number: pstn-9998 When I am calling " pstn-4444" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing ..... Answer("SIP/pstn-9998.... Asterisk should be showing "pstn-4444&quo...

...sh keys) v Ssh connect to bridge with tunnel for next ssh v Ssh connect to DST host via bridge. No direct connection SRC/DST is possible, the ssh keys only reside on base host and cannot be copied to any other host. My idea was to create ssh tunnels (plain port forward) from DST:4444 to base:5555, base:5555 to SRC:6666 (result tunnel DST:4444->SRC:6666) and run on SRC: nc -lp 6666 -e rsync --server -a . . and something like that at DST rsync -a rsync://localhost:4444/ . but that fails on src side with: protocol version mismatch -- is your shell clean? (see...

I'm attempting to set up a selenium grid on virtual machines. The default NAT is good enough for almost everything, but I need to be able to send my hub VM requests on port 4444. I've attempted to get this going by adding the following to my iptables firewall: -t nat -A PREROUTING -p tcp --dport 4444 -j DNAT --to-destination 192.168.122.107:4444 -t filter -A FORWARD -m state -d 192.168.122.107 --state NEW,RELATED,ESTABLISHED -j ACCEPT And...it's not working. I...

...e=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup=1 In sip.conf below "insecure=invite" is NOT WORKING [pstn-4444] type=friend secret=256 insecure=invite username=voice-4444 mailbox=622 context=incoming host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw nat=no callgroup=1 pickupgroup=1 Both dial plan loaded on the same asterisk using the same Audiocodes MP-114 What other variable wou...

...out waiting for the message playing) - Asterisk hangups. I'm not always able to replicate the problem but, as "Il Neofita", I'm using the italian prompts... could be a problem related to that? Bye and regards Marco Signorini. > Il Neofita wrote: > > -- <Local/4444 at servizi-463b,2> Playing > > '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language > > 'it') > > == Spawn extension (servizi, 4444, 1) exited non-zero on > > 'Local/4444 at servizi-463b,2' > > > > > It may be related...

Dear all, I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time" *CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack -- Called 1/150 -- Zap/1-1 answered SIP/4444-d365 Here are some more details to help in troubleshooting the problem Dmesg Output is as following: Zapata Telephony Interface Registered on major 196 Code test: code function addr = 0x004894f4 iR...

...ing','$_POST[pre_ring]')"; It seems that $grplist is the problem. Can someone please point what is wrong? Error: Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '('333')('4444'),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at line 3 Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/201007...

...::#1935) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465440: open_socket(0.0.0.0#4321) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465878: open_socket(0.0.0.0#2605) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465878: open_socket(0.0.0.0#4444) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465878: open_socket(0.0.0.0#8611) -> permission denied: continuing: 1 Time(s) dispatch 0xb4466008: open_socket(0.0.0.0#1935) -> permission denied: continuing: 1 Time(s) dispatch 0xb4466008: open_socket(0.0.0.0#5546) -...

...8: open_socket(::#8554) -> permission denied: continuing: 1 Time(s) dispatch 0xb4463008: open_socket(::#8614) -> permission denied: continuing: 1 Time(s) dispatch 0xb4464008: open_socket(::#8613) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465008: open_socket(::#4444) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465440: open_socket(0.0.0.0#5546) -> permission denied: continuing: 2 Time(s) dispatch 0xb4465440: open_socket(0.0.0.0#8554) -> permission denied: continuing: 1 Time(s) dispatch 0xb4465878: open_socket(0.0.0.0#2605) -...

...gest and then calculate the length of that longest element. I was hoping to find a way to simply return the length of the longest word in a more straightforward way. Short sample code -- > shadstr <- c("My string of words with varying lengths. Longest word is nine - 1 22 333 999999999 4444") > shadvector <- unlist(strsplit(shadstr, split=" ")) > shadvlength <- lapply(shadvector,nchar) > shadmaxind <- which.max(shadvlength) ## Maximum element > shadmax <- nchar(shadvector[shadmaxind]) > shadmax [1] 9 Many thanks for your help and suggestions....

...tializer element is not constant keymap.h:4442: (near initialization for `keymapInfos[6].map') keymap.h:4443: `keymap_danish_iso' undeclared here (not in a function) keymap.h:4443: initializer element is not constant keymap.h:4443: (near initialization for `keymapInfos[7].map') keymap.h:4444: `keymap_estonian_iso' undeclared here (not in a function) keymap.h:4444: initializer element is not constant keymap.h:4444: (near initialization for `keymapInfos[8].map') keymap.h:4445: `keymap_estonian_iso15' undeclared here (not in a function) keymap.h:4445: initializer element is no...

Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part -------------- An HTML

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow f...

I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS) but when call comes on pstn-4444 it goes to context "fax-incoming" in sip.conf: [pstn-4444] type=friend context=incoming ... [pstn-9998] type=friend context=fax-incoming ... the device register per end point just fine, so it can find "secret=xxx" correctly but why the call is not forwarded to correct cont...

This is the command line output: Total brick list is larger than a request. Can take (brick_count 4444) Usage: volume create <NEW-VOLNAME> [stripe <COUNT>] [replica <COUNT>] .... I am testing if a big single volume will work for us. Now I am continuing testing with three volumes each 13PB...

...ld Resource) However, when I set up a sip call between two sip phones and one end puts the call on hold, then I always get the following message and the remote side is not informed that the call is on hold: -- Executing [s at macro-stddial:2] Dial("SIP/2222-00000000", "SIP/4444") in new stack == Using SIP RTP CoS mark 5 -- Called 4444 -- SIP/4444-00000001 is ringing -- SIP/4444-00000001 answered SIP/2222-00000000 -- Native bridging SIP/2222-00000000 and SIP/4444-00000001 later when the call is put on hold: -- Music class default requested but n...