search for: devicestates

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose & debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16732which adds two new AMI commands, called "DeviceStateSet" and "DeviceStateGet". These commands let you update Custom device states, and read all devicestates from AMI. It would be very nice if someone could help me test this feature, and report back to the issue tracker. To test, log into AMI as usual, and then issue something like th...

With virtio-1, we support more than 32 feature bits. Let's make vdev->guest_features depend on the number of supported feature bits, allowing us to grow the feature bits automatically. We also need to enhance the internal functions dealing with getting and setting features with an additional index field, so that all feature bits may be accessed (in chunks of 32 bits). vhost and migration

With virtio-1, we support more than 32 feature bits. Let's make vdev->guest_features depend on the number of supported feature bits, allowing us to grow the feature bits automatically. We also need to enhance the internal functions dealing with getting and setting features with an additional index field, so that all feature bits may be accessed (in chunks of 32 bits). vhost and migration

With virtio-1, we support more than 32 feature bits. Let's extend both host and guest features to 64, which should suffice for a while. vhost and migration have been ignored for now. Signed-off-by: Cornelia Huck <cornelia.huck at de.ibm.com> --- hw/9pfs/virtio-9p-device.c | 2 +- hw/block/virtio-blk.c | 2 +- hw/char/virtio-serial-bus.c | 2 +-

With virtio-1, we support more than 32 feature bits. Let's extend both host and guest features to 64, which should suffice for a while. vhost and migration have been ignored for now. Signed-off-by: Cornelia Huck <cornelia.huck at de.ibm.com> --- hw/9pfs/virtio-9p-device.c | 2 +- hw/block/virtio-blk.c | 2 +- hw/char/virtio-serial-bus.c | 2 +-

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

With virtio-1, we support more than 32 feature bits. Let's make vdev->guest_features depend on the number of supported feature bits, allowing us to grow the feature bits automatically. We also need to enhance the internal functions dealing with getting and setting features with an additional index field, so that all feature bits may be accessed (in chunks of 32 bits). vhost and migration

With virtio-1, we support more than 32 feature bits. Let's make vdev->guest_features depend on the number of supported feature bits, allowing us to grow the feature bits automatically. We also need to enhance the internal functions dealing with getting and setting features with an additional index field, so that all feature bits may be accessed (in chunks of 32 bits). vhost and migration

With virtio-1, we support more than 32 feature bits. Let's extend both host and guest features to 64, which should suffice for a while. vhost and migration have been ignored for now. Signed-off-by: Cornelia Huck <cornelia.huck at de.ibm.com> --- hw/9pfs/virtio-9p-device.c | 2 +- hw/block/virtio-blk.c | 2 +- hw/char/virtio-serial-bus.c | 2 +-

With virtio-1, we support more than 32 feature bits. Let's extend both host and guest features to 64, which should suffice for a while. vhost and migration have been ignored for now. Signed-off-by: Cornelia Huck <cornelia.huck at de.ibm.com> --- hw/9pfs/virtio-9p-device.c | 2 +- hw/block/virtio-blk.c | 2 +- hw/char/virtio-serial-bus.c | 2 +-

Hello, my configuration is the following: a TDM400P board with an fxs and fxo daughter boards on it. I thus connect a fax to my FXS port, after having verified that this port was correctly functioning. For this, I had tried before with a simple phone, and with some basic voicemail exten scripts. Here is my simple dialplan for my fax reception: exten => 300,1,Ringing() exten =>

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the

pvSCSI backend driver Signed-off-by: Tomonari Horikoshi <t.horikoshi@jp.fujitsu.com> Signed-off-by: Jun Kamada <kama@jp.fujitsu.com> ----- Jun Kamada _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,