similar to: Problem configuring Asterisk, Please help me

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58

Hi All, I am a newbie and trying to install Asterisk from instructions given in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so I downloaded rpm's from ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried installing one by one but I get the following errors error: Failed dependencies: asterisk = v1.0.9 is needed by

Hi All, After few problems I have installed Asterisk and changed my iax.conf. I have defined a user in iax.conf and when I try to connect that user from DIAX phone I get the following error Jan 17 23:48:16 NOTICE[16448]: chan_iax2.c:3910 register_verify: No registration for peer 'manoj' (from 59.93.66.12) In DIAX phone I gave the below to connect username = manoj password=manoj

Hi All, I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new

Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:18006733555@mantragroup.com>' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause

Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP.

Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how

Hi, We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth accommodating multiple GSM frames in one packet. If we want to use per packet 10 GSM frames how to do this using asterisk? Assume the sip client is able to split these packets in to individual GSM frames. Any help will be sincerely

Hello,everyone, I encoutered some difficult with IAX when I run the asterisk. <---->internet <--> asterisk + NAT <--> DIAX my * box and NAT are at the same linux box which connecting to the internet using ADSL. The box has two network cards and two IP address,such as public IP:211.11.11.11 private IP:192.168.1.10 the windows box running DIAX lies behind the

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

Hi All, I have installed Asterisk and able to create Users and get them connected to Asterisk after authentication. My question is how can I make calls to different DIAX clients through my Asterisk server. I also have vonage softphone account, using that I tried calling 18882255322 -- Registered 'manoj' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax'

Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even

I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks,

hi i am trying DIAX and *. i cannot make calls from ne DIAX to another. whats the config? exten=>44,1,Dial(IAX/username,20,tr) ??? just a guess. any help will be appreciated. cm ===== Designs __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree

Greetings, DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop.I think), and then * goes to vmail. Here's the debug output: affinity*CLI> iax debug IAX Debugging Enabled Rx-Frame Retry[N/A]

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any

Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards