similar to: Asterisk keeps running after hungup untill I press #

Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten => 201,1,VoicemailMain(@default) exten => 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see it in the debug output of the Asterisk CLI. The call only stops if before hung up, I

Hi, I have samba-3.0.28a crosscompiled and running on a MIPS platform. The development system has about 150MB of free RAM after system bootup and no swap space. The system also has an USB interface, to which an external USB hard disk is connected. When I try to transfer huge files above (100MB) from a client on to the USB hard disk, I find that the page cache eats up almost about 100MB and

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi, I want to make some stress tests on two machines were I configured different implementations of open source sip servers. I'm thinking about making some graphics like CPU and memory usage extracted by SNMP while flooding my servers of sip calls. Does anybody know some good software to do that? Regards, Ricardo.

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in

Hi, Has anyone managed to create a setup with a Sipura SPA-1001 as a client? Right now I can connect to the device by dialing the extension number but when I try to connect from the phone handset to make an outbound call it gives an unavailable tone. I'm using Line 2 on the SPA-1001 to connect to the local asterisk server, line 1 is used to connect to my VOIP provider until I can get the

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

Hi, I guess something's wrong with my chan_dahdi configuration, ... but I can't seem to get it. When I test incoming calls on a DAHDI-channel (incoming from pstn), asterisk seems to interpret it as a caller hangup after each ring. Any ideas. OUTPUT: -- Starting simple switch on 'DAHDI/5-1' -- Executing [s at from-pstn:1] *Verbose*("*DAHDI/5-1*",

Hi. [I asked a similar question a while back, but unfortunately wasn't around to reply to the responses, so sorry if you experience any deja vu.] I have a * server acting as an IVR system. The calls come in via IAX. After a period of about 40 seconds of silence (either waiting for the caller to dial an extension, or with the audio paused in controlplayback), the call hangs up. All I see in

Hi, I'm tying to delay the HungUp. I tried this way: exten => h,1,NoOp(Start) exten => h,n,Wait(5) exten => h,n,NoOp(End) exten => h,n,Hangup() but it doesn't work, Any idea? Thanks in advance.

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've got a system set up like: POTS <-----> Asterisk 1 <-----> Asterisk 2 <-----> IAXComm The POTS line is connected to "asterisk 1" via an X100P card and "asterisk 1" is connected to "asterisk 2" via ethernet. With incoming calls from the POTS line, everything works, but between 5 and 10

I installed this card, everything work, i can make call and receive call with no echo and great sound quality, but after between 5 to 50 secs the call disconnect by itself, in the log i don't see nothing revelant. I don't share any IRQ, zttest show me values between 99.98 and 100. The only thing i see, the pstn line is not realy a true pstn line, it's plugged in an Arris Telephony

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes