search for: myphon

>> Somehow sysvolcheck is using a LOWER CASE 'f' in the GUID folder name >> for the default GPO! >> >> Where is this coming from? Of course, in Windows this doesn't matter. >>> But in linux it is a showstopper. > I think it is coming from AD. > You will probably have to rename the GPO in AD, possibly along with the > 'name'

...my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2002] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes ;******************************* ; BEGIN: CUCM(s) Added Bel...

...nce with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI> == Using SIP RTP CoS mark 5 -- Executing [6003 at myphones:1] Set("SIP/6001-0000000c", "_SIPSRTP_CRYPTO=enable") in new stack -- Executing [6003 at myphones:2] Dial("SIP/6001-0000000c", "SIP/6003") in new stack == Using SIP RTP CoS mark 5 -- Called 6003 -- SIP/6003-0000000d is circuit-busy == Everyo...

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536 handle_alarms: Detected alarm on channel 5: Red Alarm -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' == Everyo...

...Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb) exten => _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor) exten => _X.,n,Dial(SIP/myphone,300,tTo) My [applicationmap] in features.conf is setup as follows: in-pauseMonitor => *7,self/callee,Macro,pause-record in-resumeMonitor => *9,self/callee,Macro,resume-record I also have the following contexts setup in extensions.conf: [macro-pause-record] exten => s...

...al string just seems to prevent the call from going through. Username is your 10-digit phone number. Password is in the .reg file they sent you via email. I signed up for the USB phone, so I don't if they send a .reg file if you went for the MTA. sip.conf register => 1234567890:password@myphone.voiceglo.com [myphone.voiceglo.com] type=peer username=1234567890 secret=password nat=no host=myphone.voiceglo.com disallow=all allow=ulaw allow=alaw canreinvite=no qualify=200 restrictid=no fromuser=1234567890 fromdomain=1234567890.voiceglo.com extensions.conf exten => _91NXXNXXXXXX,1,Dial(...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no answer) to a specific number on * (5901) that is my x-lite software client. If 5901 is

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

...nfig as small as possible to help the troubleshooting process. Attached is he most recent debug. My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251 SIP.CONF [6001] type=friend secret=cisco2003 callerid="Dave" <6001> host=dynamic canreinvite=no context=myphones regexten=6001 [CM8] type=friend host=10.169.169.250 canreinvite=yes ;disallow=all allow=ulaw allow=alaw qualify=yes context=myphones Extensions.conf myphones] exten => 6001,1,Dial(SIP/6001) exten => 6001,2,Hangup() exten => _X.,1,Dial(SIP/CM8/${EXTEN:0},30,rt) Thanks for any he...

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

...e is played exten => 202,n,Playback(vm-goodbye) ; Hangup() ends the call, hangs up the line exten => 202,n,Hangup() After loading extension and dahdi, i called from jitsi and dialed 81 but asterisk is giving o/p as below and busy tone is coming on jitsi -- Executing [81 at myphones:1] Dial("SIP/sandeep-00000000", "DAHDI/1,20,rt") in new stack -- Called 1 [Nov 2 14:45:31] WARNING[2145]: chan_dahdi.c:7536 handle_alarms: Detected alarm on channel 1: Red Alarm -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Everyone...

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

...called identapop pro. I havent tested incoming calls yet but so far, I cant get calls out using outlooks. I configured TAPI for asterisk inside outlooks and I set TAPI to these configs: TAPI connects using the manager to asterisk without problems. As channels I configure this: User channel: SIP/myphone and the phone actually rings when I tell outlook to dial out. Inbound channel: Zap/g1 but I havent tested this yet, please let me know if it looks ok? I have a group defined as 1 wichi has 2 x100p cards. Dial by context with nothing defined Outbound channel: I use to have Zap/g1 but nothing happen...

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as...

...on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): =========== exten => s,1,AGI(netcid.py|${CALLERID(num)}|${CALLERID(name)}) exten => s,n,Dial(${MYPHONE},5) =========== # cat netcid.py #!/usr/bin/python import socket,sys,time,os def sendstuff(data): s.sendto(data,(ipaddr,portnum)) return sys.stdout = open(os.devnull, 'w') if os.fork(): #BAD? sys.exit(0) os._exit(0) else: now = time.localti...

...ting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes echocancel = yes echocancelwhenbridged = yes relaxdtmf = yes rxgain = 0.0 txgain = 0.0 group = 1 callgroup = 1 pickupgroup = 1 immediate = no context = myphones signalling = fxo_ks [phone1](phone) signalling = fxs_ks callerid = "Andrew F Robinson" <(503)543-2338> dahdichan = 1 [phone2](phone) signalling = fxs_ks callerid = "Michael C Robinson" <(503)987-1322> dahdichan = 2 [phone3](phone) callerid = "2010" &lt...

...client, and also forget everything/everywhere about ports below 445. > > Provided that: > A) Samba running on termux works as it does on Linux > B) It is using SMBv2 > C) Your camera users SMBV2 > > Then the following smb.conf should work: > > [global] > netbios name = MYPHONE > ... > with SMBv1, removed. > > I suggest you start by tryin... ... > If this all works, then disconnect and try to connect from the camera. > > Good luck > > Rowland I'll check all this when I get out of the surgical block if I ever get awake ;-) I'll try with...

I have tidied up the script and added some help text, feel free to download and maybe improve. http://www.masonc.com/phoneconf Usage: Usage: ./phoneconf [config|help] phonemacaddress extention username context ./phoneconf help will print syntax info ./phoneconf 0004f201aa11 500 MyPhone defaulot will configure the phone with that mac with one extension (500), and add the phone to sip.conf. Edit at least the first two variables before using and change the sip.conf stuff to suit, this works well for me on internal and remote phones. Improvements greatly appreciated, I'm a c...

...s sent by Asterisk. The problem is Asterisk is unable to recognize any DTMF tones I enter on my mobile, thus I'm unable to do anything with the dialtone (hangs up after 10 seconds). However - if I change the [CallBack] context to; [CallBack] exten => s, 1, Answer exten => s, 2, Dial(SIP/MyPhone) My local SIP phone rings and I'm then able to hear (and transmit) DTMF tones from both end points. As a sidenote, I can make calls in and out the provider and DTMF works fine. Does anybody have any ideas - any help is appreciated.. Cheers Joe

hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten => s,1,NoOp() exten => s,n,Dial(SIP/120)