similar to: MediaHandling--Help Required

Hello list. I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm using two TC400B cards for transcoding. When I reach nearly 100 simmultaneous calls, the CDR radius packets are being duplicated and I'm getting this message in the asterisk console : cdr_radius.c:227 radius_log: Failed to record Radius CDR record! I'm also using the radiusclient-ng 0.5.6

Sorry, but why u r using the Radius with the CDR? Not enough to access the CDR in the /var/log/asterisk/cdr-csv/Master.csv? Also, what kind of Radius u r using? Any suggested link? Regards Bilal > > Hello list. > > I'm having some problems with the CDR Radius in my > Asterisk 1.4. I'm > using two TC400B cards for transcoding. When I reach > nearly 100 >

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

Hello all, I am testing the Dial application with the fall through priorities for different cases what i want is the flow after failure of the Dial application which simulates response codes like 1)404 -- Not found 2)480 --Temporarily Unavailable 3)486 --User busy i did manipulate the priority flow like the following for the case 2 and 3 ... exten => _XX,1,Dial(SIP/extension) exten =>

Hello all, I am struggling with sending voicemail as an attachement in Email. When i have given the email like someone at gmail.com it is delivering to my gamil account perfectly(of course to spam folder). But when i given the email like someone at mycompanymail.com it is not delivering to my company email account.. What should i do ? Actually my company is using a third party email server..

Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hi, I'm using Asterisk 1.2.3 with the asterisk-oh323 channel driver, version 0.7.3. Pwlib is V1.8.7 an OpenH323 is V1.15.6. Following CallFlow: SIP-UA -> OpenSER -> * -> CCM OpenSER routes all calls with prefix 60 to Asterisk, where I've configured following extension: exten => _60.,1,Dial(OH323/${EXTEN:2}@v.w.x.y) v.w.x.y is a Cisco Callmanager where Asterisk is

Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001

Hi, Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. Recently I have been getting errors from Asterisk due to corrupted From: headers, which appear to be caused by uac_replace. Here is a section of the debug log: Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: removing <From: <sip:lenc_domain.com@sip.domain.com>;tag=635c3ce6 > Mar

Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1->OpenSER->Asterisk->Openser->UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and

Good afternoon everybody. I first would like you to excuse me for my english. I have an issue with a SIP call ID which is not changed in the call configuration described bellow : I have an Asterisk Server A using only SIP protocol. Behind A there are 2 distant clients (using softphone X-lite) C1 and C2 and a proxy server OpenSIPS (ex OpenSER) P. The idea is that when C1 want to call C2,

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Hello users, have been facing problems with t38 passthrough using asterisk 1.6.0.3. observed also that in case of SendFAX we are not having major issues, its almost successfull. ReceiveFAX has problems most of the time. we have been using a ringcentral account for testing this receivefax. so ringcentral is trying for 3 times if the sending fax failed for the first time. what i observed is

Hello users, i have successfully installed the cepstral voice and in the text only mode its working fine when i swift applicaiton in dtmf mode like exten =>111,1,Swift(hello user| 5000|1) exten =>111,n,NoOp(dtmf is ${SWIFT_DTMF}) exten => 111,n,Hangup() case1: when i am listening to the hello user prompt if i press any key 1,2,3,4,5,6,7,8,9,0,*,# i am getting the ${SWIFT_DTMF } value

Hello all, Can anyone have any experience working with service provider like Talkfree . They are giving the user accounts based on the single user accounts and those needs to be directly register to the service provider not to the local system i have taken a connection which when configured to service providers domain direclty ,xlite can make calls without any problem but if i want to use it

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Dear ALL: I find a program named "asterisk_b2bua" on http://developer.berlios.de/projects/b2bua/ And I also download them(two components) and try to test it. But I have not enough knowledge about asterisk. It seems a Software PBX. Does asterisk_b2bua work? Does anybody ever try it? I have questions about my scenario. |======================> UA2

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method==