search for: adamk

Hi List, i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs -> me -> deverto4) the call setup is complete but there is no audio. I can see the audio in the form of

Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part

Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP

Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same "bag" of minutes. Meaning ext 101 use 1900 mins, ext 102 60 mins and ext 40 mins. The limit must be monthly. I guess some "billing" solution can do the trick,

Hi List, i've been receiving several sip registration probes in the last month, and as this server is a testing site (no external lines, no nothing) i have no fail2ban and still not planning to install. Whenever i have nagios telling me that there is another 'guest', i go and edit iptables manually and that's it. Recently i discovered that these attacks start with some kind

Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R ? stops moh and rings once an agent is ringing (Asterisk Trunk) (in

So I've spent the last few days getting Xorg pulled from freedesktop git and built. Everything seems to have worked, and I now have Xorg version 7.1.99.2 running, composite enabled, and AIGLX enabled. The Xorg log file shows: (WW) AIGLX: 3D driver claims to not support visual 0x23 (WW) AIGLX: 3D driver claims to not support visual 0x24 (WW) AIGLX: 3D driver claims to not support visual 0x25

...; > Thanks for your help again. Would be nice really, but I'll try to find a > workaround to avoid that problem (or ignore it). > > Bjoern > > > > > ------------------------------ > > Message: 2 > Date: Sun, 10 Feb 2008 18:44:46 +0100 > From: Adam KOSA <adamk at 3a.hu> > Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco > pix 506 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <47AF380E.9090109 at 3a.hu> > Content-Type: text/plain; cha...

Hi guys, sorry for the long e-mail, i'm only trying to give as much information as i think is relevant to my problem (console log, sip.conf and extension.conf parts). i've been practicing with callback for a while, but i'm at a dead end. I hope somebody can help me to move on. i have troubles getting two calls bridged together. Scenario is the following: - asterisk calls my

Hi folks, my goal is to access voicemail (there were some posts about this) but not by dialing numbers. As asterisk sends voicemails in e-mail, it's cheaper for us to read e-mails on our cell phone (3g, gprs), and the message is attached there. i've looking around in voicemail.conf and found: [general] ; Default formats for writing Voicemail format = wav49|gsm|wav my phone

Greetings, i've been posted a message to this list in july, which had one response. Thanks for that idea! Unfortunately asterisk is only a hobby, and did not have much time dealing with the problem since. My original letter was long, i wouldn't post it again, the archive url is http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html Since than i've upgraded to

Hi, i've just finished setting up gtalk connection with asterisk. it works nice, audio is full duplex. i just have one question which i could not find an exact answer to. Is gtalk able to send dtmf codes? Because i'd like to listen to my voicemails while away from home. I've been googling for half an hour, i found some sort of jingle protocol which i'm not sure what to