similar to: Forward DID to another server

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

dear all, what is the meaning of this *Got SIP response 603 "Declined" back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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'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

hello, Does anyone know about setup Message wait indication between asterisk and kamailio my phone are registered on kamailio and voicemail leaves on asterisk server. how do i notify to kamailio that 1 message is leaved for you on your mailbox. and also i tried all script listed in voip-info.org. any one know any working method or anybody have some type of setup which may help me any help

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

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

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

Dear All, I want to configure Asterisk/Kamailio Like system monitor with Heartbeat is there any way to monitor Service If NODE1 is stopped or over loaded then NODE 2 will work and vice verse. any help appreciated because i m stuck in heartbeat to configure service. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

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

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

Really? Me? Oh Pavel! I would be inestimably honoured. On 04/27/2012 01:55 AM, Pavel Ismailov wrote: > Hello! > > My name is Pavel Ismailov > and I`m CEO of www.flashphoner.com project. > > We noticed that you quite active in Asterisk-user > mail list, and would like to offer you buy signature > in your messages for some monthly price. > > Is it interested for

Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected.

I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of

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 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 am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090712/b5bab36d/attachment.htm