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...a sip invitation problem with my voip provider and here the message that was shown : Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:XXXXX@195.112.214.99>;tag=as7b43dfbd' -- SIP/callshop-3fcc is circuit-busy == Everyone is busy/congested at this time -- Got SIP response 481 "Call Leg Does Not Exist" back from 213.61.187.150 My sip box is :195.112.214.99 The voip provider sip box:213.61.187.150 the configuration of my sip file was like this:...

i have opened an account with callshopcompany,and when ive tried to send calls by the sip i had a message show an asterisk invitation problem i had these sip configuration: sip.conf [callshop] type=peer host=sip.callshopcompany.com username=XXXXX secret=XXXXXXX Then i tried to add these lines and it worked :...

Hi, Can someone explain to me how ATAs operate in a call shop type environment to provide realtime billing to the callshop software. The ATAs seem to be configured to connect directly to the asterisk server, so how does the call shop software report in realtime your destination and duration of the answered call? Thank you, AC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists....

Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance. Dan -------------- next p...

...o I need to ever compile and load codecs modules in the firmware built ?? space is limited in the board. Do i need to compile the "format" modules too ?? must I put allow g729 in the confs? on the other side I need to generate CDRs, one in the board for " sale" rates in the callshop an other in the Lucent compact switch for "cost" rates, question is. I need to answer in the board before dial iax2 because of sendtext parameters, then in the dial iax2.. Do I need to reset cdrs in order to do not count time between "answer" and far end answer ?? At the oth...

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I