similar to: How to connect two asterisk server

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

hi all, i need to integrate and modify one of the application in asterisk/apps section... whenever i modified small steps ..in order to check and compile i 've to do recompile the whole asterisk module and it consuke to much time... please anyone couls you tell me, how can i modify it , compile and test the I/O in asterisk applications in a easy way... plz do reply .. Thanks for ur

Hello list, I have prepared a couple of new tutorials you may find interesting: - Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216 - Installing the Digium's Asterisk GUI for 1.4 - at http://astrecipes.net/?n=217 It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI. Any comment is welcome - the site is a wiki, so feel

Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Can anyone in the list help me with these queries on Asterisk Business Edition. *1. Why would anyone choose the Business Editon when the whole thing is avalable as GPL?* ** *2. Is there a GUI to manage asterisk?* ** *3. Can it be compared with Asterisk NOW?* ** *4. Is the CD a complete installation package?* ** *5. If im looking for hiring a server on a remote location how will i be able to

Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and

I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just

HI all, Any of you having experience with voice master? I tried using the openh323 channel it doesn't give me voice at all. THere's no packet coming in. There's no problem with any other equipment but voicemaster doesn't send voice at all. Funny thing, i have an old version of OpenPhone, it's working. So please if any of you knows this problem, please share. THx a bunch

Hi Thiru - > Could u tell me ,how to connect 2 asterisk server using sip as a > clients... > asterisk server are in same network... You can connect them either as "friends" or as "users/peers". I generally recommend the user/peer method for connecting two servers since it clearly delineates which codecs and contexts are allowed. Your sip.conf files will look

Hi , I would like to assign a Network Interface card to Physical Slot X in the virtual machine, I have gone through the XML file definition[Now i have some idea on how to position the Network Interface in Logical PCI Slot.] I am NOT interested in using a PCI Pass-through from the Host. currently running Fedora 19 with libvirt 1.2.0 I would like to create an network interface card on

Hi Laine, Yes we are setting the names as GE0-0 manually. We have turned trust ON for host IGB driver. [root@nfvis libvirt]# ip link show GE0-0 3: GE0-0: <BROADCAST,MULTICAST,ALLMULTI,UP,LOWER_UP> mtu 9216<tel:9216> qdisc mq master ovs-system state UP mode DEFAULT qlen 1000<tel:1000> link/ether a0:23:9f:ce:b1:f8 brd ff:ff:ff:ff:ff:ff vf 0 MAC 52:54:00:29:3c:be, spoof checking

Hi moises, Hi i need to done some modify/changes in one of the asterisk c source code ,,eg: app_meetme.c How can i compile and debug it without compile the whole module of asterisk... and also let me know which editor suitable it ,I'm using suse linux.. Plz help me reg. this .. -nsthi, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Help needed Currenly i'm running samba on a RedHat 7.3 system. The primary domain contreller in the network in a windows 2000 server machine. I'm using the windows 2000 server to authenticate my users in the domain. Each time a new user is created in windows 2000, i have to create the same user in linux inorder to allow the user to have access to the samba shares. Is the re a script or

The following is mainly a capture of parts of multiple off-line discussions within members of the Crossbow team (Gopi, Thiru, Roamer, May-Lin, Thirumailai, Nitin, KB, ...), I thought I''d open it up to other participants. Crossbow''s core scheduling involves switching a NIC (or individual Rx rings on the NIC) to polling mode. The receive interrupt will become not only rarer,

Hi Laine, Thanks for your detailed response. I have replied Inline. Thanks Thiru. -----Original Message----- From: Laine Stump <laine@redhat.com> Sent: Sunday, May 6, 2018 3:46 PM To: libvirt-users@redhat.com Cc: Thirunavukarasu Sengalvarayan -X (tsengalv - HCL TECHNOLOGIES LIMITED at Cisco) <tsengalv@cisco.com>; Chanda Mendon (cmendon) <cmendon@cisco.com> Subject: Re:

Hi Laine, Thanks for taking the time to respond to my question. I think I have not described my problem clearly. Let me explain my issue below with the information that you had requested. My assumption according to the information you gave me is that the admin MAC and VF MAC are the same in my case. I see a PF (GE0-0) interface but I don’t see a vfnetdev interface as you mentioned in your

Is it possible to call a worker from inside a worker? Right now, if I try, I get a recycled object error.

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.