similar to: iptables miss up phone calls if not used properly

Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a

Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hi Guys, i am using the following config in pbx1: register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176> [pbx1] type=friend

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC

Hi Guys, do any body know how to receive debug info on RxFAX application? i am experiencing a lot of fax failures and can't guess the reason behind. Thank you very much for any help! -- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100402/cf26573a/attachment.htm

Hi List, I did follow the procedure to install Free Fax for Asterisk successfully till i came accross this isssue: i can't load the fax module: pbx3*CLI> module load res_fax_digium.so Unable to load module res_fax_digium.so Command 'module load res_fax_digium.so' failed. [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as

Hi Guys, Do you know of any asterisk community version that does video codec trans-coding or in other words supports video? I have 1.8.8.1 and I see h263.c format files but can't see that codec in make menuselect. it might be just a license issue (if h263 has to have license), but not sure if community versions offer video calls at all. Thank you! -- Khalid Touati -------------- next part

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

Hi all, I am running to the following problem, when using the below dialplan to receive fax, everything works perfect till this line exten => receive,n,ReceiveFAX(${FAXFILE}): and then the following line cannot be executed, it's like asterisk can't go back to dialplan and continue, the good news is when i check what is received in my fax folder i find that the file is a valid one (not

Are there any updated instructions for installing Windows virtio drivers on a KVM Windows XP vm under CentOS6? The virtio .iso directory structure is slightly different from what is described in the RHEL6 virtualisation manual, and when I attempt to install e.g. the viostor driver, the device manager cannot find anything better than what is already installed. I tried each of the

Hi Guys, i've connecting two pbx server successfully for several times using the following config: register => USPBX:mypass at 122.11.176.35 <USPBX%3Amypass at 122.11.176.35> [PBX1] type=friend host=122.11.176.35 trunk=yes sercret=mypass context=external deny=0.0.0.0/0.0.0.0 permit=122.11.176.35/255.255.255.240 insecure=very allow=all nat=yes qualify=yes canreinvite=no in the other

Hi Guys, i could turn debug on in a asterisk 1.6 box (by enabling debug in logger.conf and core set debug to > 0), but my issue is i cannot enable debugging in a 1.2 box by doing the same 2 steps, also this is a production server so i can't restart with debug enabled, do you guys know how i can turn debug on or just know why it's not getting enabled? Thanks a lot for your help! --

Hello, I am new in this mailing,please if somebody knows how to make kdump able to load in startup. i will be very thankful, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20090409/d3435a68/attachment-0002.html>

Hi All, i have a question, is there any soft-phone available for Black Berry use, I've been told there is a firefly one, but when i looked, i found nothing, is any body has an update on this please? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/1bf1a72e/attachment.htm

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats

Hello again, I was able to answer my own question after studying the documentation and scratching my head a bit. Here is the way I did it: Multiple Subnet statements are allowed in the host file. I added a Subnet for Net C in fwall B's host file Added firewall rules to support traffic from Net A to Net C Added a network route in the default gateway on Net C so he knows where to send packets

Hi, I should have win32-changenotify and win32-semaphore out this weekend at some point. Then, the next version of win32-file. And after that? I don''t know. What do you think we should tackle next? What interests you? Different mutex flavors? IE controllers? Exchange interfaces? I think we should probably look a little more at what Python has going forward, too. Just a

As I explained, I want to be able to run a second instance of DC through another port to do testing while not interrupting the users using the primary instance in the the regular port. I was instructed to create a second config file with different listen ports and a separate base_dir. I did so and invoked dc with -c pointing to dovecot2.conf. Jun 7 10:04:23 mercury mail:info dovecot: