similar to: SIP REGISTER

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | |

Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:

I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: > /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi > install_driver(mysql) failed: Can't load > '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' > for module DBD::mysql: libmysqlclient.so.15:

Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am having a very strange problem. I am terminating a PRI (5ESS switch type, national plan, 23B+1D (24)) into a Sangoma A104 and am not able to produce any audio heard on the PSTN end of the call. Not sure what's wrong - the card worked before under a Trixbox setup. I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as wanpipe stuff would not compile), zaptel

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route

Hi folks! I have a problem with our Sip provider that have a Softswitch Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working with G711 with them. They start sending calls to our pbx, some time after they start to receive 408 messages from asterisk and some time after this they start to complete calls normally, I don?t know what can be wrong. Someone has configured asterisk to

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome!