similar to: duration sec and billing sec in cdr

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

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very

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

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi, Is there any reason against making a ctdb connected 2 pc samba cluster also a domain member? After setting the [include = registry] option, one member of the cluster didn't let the users to log in. If I relogin this cluster member to the domain, then the other member starts to refuse users to log in. Did I miss some option that I have to use in this scenario? The name of the servers are

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

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

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 all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

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

Hello! Hardver: Windows XP Compaq Proliant DL360, Linux Compaq Proliant DL380. 2 pieces of processors Intel Xeon 3,2GHz, 2GB RAM, 6 gigabit interface (2 tg3, 4 e1000), Debian Woody, 2.4.31 vanilla kernel. 6 U320 SCSI 15krpm HDD, 2 HDD RAID1 system, 4 HDD RAID1+0 data. Every network cards connect at speed of 1000MB full duplex, with XP crosscable (but we've tried with gigabit switch, too, we

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