similar to: Dial problem with Asterisk 1.8.4.4

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Had my Asterisk instance stop responding to incoming/outgoing calls today. Had to kill -9 the asterisk process and restart it to get it back. Not really looking for help on that as the instance is version 1.6 and is due to be replaced with an upgraded version shortly. However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until

I am trying to get something working that is just not doing quite what I want. It may not be possible, but I figured it was worth asking about. The details: Asterisk 11.6.0 Polycom SoundPoint IP650 phones running 4.03 firmware. We have a queue with 4 phones in it. ringinuse is set to yes and the stategy is ringall. In sip.conf, we have notifyringing set to yes as well. Asterisk is sending

Hi! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

Ok. Thanks for the hint. But, what exactly is a "System() dialplan application"? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________________ De: asterisk-users-bounces at lists.digium.com

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over

Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten => 1234,1,System(echo "This is a test" >> /var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain <darcy at vex.net>: > On Mon, 8 Jun 2015 22:24:33 +0200 > Luca Bertoncello <lucabert at lucabert.de> wrote: > > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > > > Basically, they are hoping that you are running the equivalent of a > > > mail server open relay. They are trying to use you

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Make sure you have solved the problem. You don't want to get hit with a > phone bill for calls from your location to Israel. Basically, they are > hoping that you are running the equivalent of a mail server open relay. > They are trying to use you to dial out to another number. You don't want > to pay

I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this

Hi All, We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue (maqueue) structure for handling customer calls. There are 4 queue members (85744,85766,85511,84888). These 4 members are logged in using AgentCallbackLogin application. But at some point, one of the agent's SIP phone does not ring for an incoming call to this queue. I checked the agent status and it is not in paused

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Thank you Kevin, I've looked at your solution and while I agree it's not ideal it does appear to be something that might work for me. I'll see if I can maybe backport the QUEUE_MEMBER stuff to 1.8 from 11. I'm also exploring an idea with a co-worker of using an AMI listener that will fire off actions in response to the member being paused and doing things that way. I looked at

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Based on SIP packets coming in from IP addresses you don't recognize, > while you may not be hacked, you would seem to have people probing your I think, too, it's someone probing my IP... > system. One thing you can do at the firewall level is restrict inbound sip > communications to only those from your

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. Here is a link to pastebin with the Sip trace. In it you