similar to: clarification about bridge the call

? Hello all, We are setting up a gateway in which the SIP devices will be connected dynamically using the Asterisk system. We use the originate Manager API command from our code to call an IP as (SIP/1@10.20.30.40). The call rings on the phone and goes through the normal (default) context and finally hangs up(WARNING[13833]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context

Hello all,   I am using HALF DUPLEX modem for TAPI call.the following message is displayed while i am starting the AsteriskNOTICE[1416] chan_tapi.c: Channel format set to ULAW\' ERROR[1416] win32_tapi.c: TAPI Error: 80000023 (HCALL 0x0) on lineGetID . If i will receive an Inbound call to modem, i will answer that call............and put an wait for infinite time.The

Hello sir: There are 2 questions about string. 1 How to calculate the width of a string? e.g string "abc"'s width is 3; 2 How can I get the "substring" in such kind of condition: "f:\\JPCS_signal.txt" "f:\\PC1_signal.txt" "f:\\PC2_signal.txt" What I wanna get is "JPCS" "PC1" "PC2".How can I achieve them by R

Hello all! I'm new in this ML and I write you for a suggestion about integrate Asterisk and SER. My idea is to use Asterisk as a local PBX server where users can authenticate and make local calls, but when a user dial a non local number, an asterisk extension call SER Server who redirct to right remote asterisk. Originally I make this only with asterisk where in everyone I setted iax.conf to

Hello everyone, I have a issue about the KVM block migration. Please give me some help. 1) I use the "virsh create" command to start a KVM VM in source machine. 2) And then, I use "virsh migrate" cammand to start a block migration: # virsh migrate --live --copy-storage-all --verbose win7 qemu+ssh:// 186.100.8.136/system root at 186.100.8.136's password:

hello all,i will make a call to asterisk server, that time the end user in ringing phase.After that i am trying to \"redirect\" the call during ringing phase.This time the server shutdown...............i want to answer the call during ringing phase.please help me if anyone knows.Regards,Pandi.P -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Can someone tell me how I could redirect from an url like "www.url.com/mycontroller" to "www.url.com/en/mycontroller"? I was reading all sorts of routing documentation but couldn''t come up with a solution. Of course this should be a general rule, so "www.url.com/mycontroller/myaction/myparam=99" should naturally redirct to

No. No solution identified. I'm just closing the log and process windows and minimizing the X11 window. Annoying, but not a show stopper. You're using a later version of Wine than I am; I'm sorry to see it hasn't fixed this. On 7/8/08 wine-users-request at winehq.org wrote: > Message: 7 Date: Mon, 7 Jul 2008 23:10:35 -0700 (PDT) From: ewans > <esadie at

Hi, I use OpenSSH (thanks folks) to administer a mix of boxes at work and have seen some quite scary problems. I set up an ssh connection from the host to a central admin machine from a perl script running on the host. The script brings up an ssh connection to the central admin machine and sets up some reverse port redirection for administration (telnet for instance) and some forward

Hi all, I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of command DIAL when used from the dial plan. Can some one guide me how to get the vaue of

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

I am having a problem interacting with a remote IVR system when the outbound call is going via SIP. The only way that I have been able to get a response from the IVR is to set dtmfmode=info in sip.conf. Unfortunately that doesn't quite fix the problem because it will still only accept DTMF input once the voice response has finished on the IVR. If I try and press anything while the IVR is

Hi, Im trying to do this: 1) Originate a call between an external number and a ivr that do some things in background 2) after the originate I bridge the person that dial that extent with the external number I would like to have the ivr in background while the bridge is up for monitoring porpoises, but seems to stop processing when the local bridge is done other possibility could be

Hi folks, in 2.4 kernels, device matching for bridged packets was done with iptables -i/-o. Since 2.6, I was used to use -m physdev here. In 2.6.18, This seems to be more complicated. At least the filter/INPUT chain now doesn't match with -m physdev --physdev-in anymore, but FORWARD and OUTPUT does. I also read the note that -m phydev is now deprecated for non-bridged traffic. Does this

Hola Knut, Thanks for your quick answer, but still having the same error. I have created the following users: (bppr-r40) IBM AIX Server: username: s670587 SAMBA SERVER (Above Server): smbpasswd -a s670587 WINDOWS 2000 Client: username: s670587 Have set all the passwords to 'manager', but still getting the same error message when I try to do a MapNetworkDrive

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Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip

Hi people! Sorry for bringing up some annoying issue.. yes, it's AMD again... But I was searching the last days for a solution for my problem and didn't really find anything. Now I'm hoping that someone of you has maybe an idea for me. :) My setup: --------- I use the Asterik Manager API to generate outgoing calls (by using "Originate" messages). These outgoing calls