similar to: Call Forwarding/Redirection?

Hi, We recently got a PRI installed at one of our local centres however, we appear to be having interesting issues. Everything seems to be installed in the correct manner. asterisk*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203

Hi, We've got a customer site with Cisco 7940G Phones connecting to our Asterisk Server via SIP. However, upon running a "sip show peers" it states that the phone is : user/user (Unspecified) D 255.255.255.255 0 Unmonitored All phones can call out via our Asterisk Server fine regardless. The problem is when they try and place a call to one another, where it

Hi there, Anybody on this list knows where I can obtain Hong Kong DID's from ? Cheers, Sahil

Hi, On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio. My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx "pri show version" libpri version: 1.4.12 A

Hi folks. My client is wanting to use call forwarding configured on their phones (Linksys SPA942), with a PRI from their provider. When we configure call forwarding, we invariably get a "The number you have dialed is not in service" message from the providers. Examining the detailed dial plan debugging as well as the PRI debugging, the number is dialed correctly. The only difference

Hi sahil: I think the reason why you cannot get page shared is due to the gref you got. Gref is responsible for a page allocated from domU, in my understanding it should not be 0, that is a gref 0 can not be shared, that''s why I skip gref 0 to be nominated. The gref is nominated to Xen and later used to find a corrspond MFN, so it shall not always be the same.

Thanks For support. One more doubt is that , As Centos 7 major release is launched , now support for centos 6.x where x is latest mirror point release is supported or not ? ? and further 6.y where y>x will be released or not . You are requested to calrify . On Fri, Aug 1, 2014 at 5:30 PM, <centos-docs-request at centos.org> wrote: > Send CentOS-docs mailing list submissions to >

Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

Roberto estoy en uruguay en estoos momentos. Recien lllego el miercoles El mar 31, 2013 1:59 p.m., <asterisk-users-request at lists.digium.com> escribi?: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users

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On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

I have dovecot and postfix set up on debian lenny. Squirrelmail is used to access the mail. I put a .forward file in my user's home directory, and that sure gets the job done of forwarding email, but I need a copy saved on the local machine as well. How do I do that?

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.2 DAHDI-Tools-v2.9.2 dahdi-linux-complete-2.9.2+2.9.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Mostly stability patches affecting

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.2 DAHDI-Tools-v2.9.2 dahdi-linux-complete-2.9.2+2.9.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Mostly stability patches affecting

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.2-rc2 DAHDI-Tools-v2.9.2-rc2 dahdi-linux-complete-2.9.2-rc2+2.9.2-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release is

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.2-rc2 DAHDI-Tools-v2.9.2-rc2 dahdi-linux-complete-2.9.2-rc2+2.9.2-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release is