search for: myphones

...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 Below...

...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 == Everyone...

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' == Everyone...

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. 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)

The call quality is really pretty good. I think better than Vonage over an FXO bridge. If you are looking for a home provider with direct SIP support and local phone numbers this is a good choice. If anyone has questions or comments about my configuration please pass them along. I have noticed that if you don't put fromuser=phone# then the extension caller id passes through. Also the

-----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 help...

<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 i...

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

I just installed tapi and some app 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

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 timeout value,

Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up 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 =>

...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" <2...

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.

Hi I'm trying to implement a Callback mechanism whereby I generate a Call file and connect an arbitrary extension with my cellphone (via a SIP Channel). If I create a .Call file that connects the channel "SIP/12345678@Provider.net" with a local extension/context I get some weird issues with DTMF tones. I've set dtmf=2833 and the codec in use is G711a. For example - I create

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)

Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - hunger at sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 D?sseldorf HRB

Scott Robert Ladd has realized an analysis of GNU C and C++ optimizations, using a genetic algorithm to discover the most effective optimization flags for different algorithms. Check it out: http://www.coyotegulch.com/acovea/index.html AMD Optimized Windows XviD codec: http://net314.myphone.gr/xvid_amd.html Happy coding ! <p>>>Forward Agency In progress we (always) trust.