We have a system that interoperates with an external service, so that the basic call flow is:PSTN origination -> Asterisk A -> External service -> Asterisk BInitially the SDP from the external service tells the two Asterisks to send RTP directly to e..
Author : David Cunningham
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We have a problem where Asterisk is resetting the CSeq on a re-INVITE, and the phone receiving the re-INVITE is rejecting it, probably as a result of that. Would anyone be able to offer any insight please?The scenario is:Phone A makes call 1 to Aster..
We have an Asterisk 13.38.2 server which today started giving we couldnt allocate a port for RTP errors. The output of netstat -anp showed that Asterisk was using all 10,000 ports allocated for RTP, even though it only had a maximum of around 200 concurr..
We have an Asterisk dial which sends the call via a proxy using //, for example:Dial(SIP/${EXTEN}@peer_address//proxy_address)Does anyone know how we can make the SIP to the proxy use TCP? We tried making proxy_address match a peer in sip.conf with transp..
We have an Asterisk server with 3 IP addresses, and need to listen on only2 of those. This is with chan_sip. Does anyone know if its possible?If Asterisk listens on the third address then it seems to cause problems with the media address put in the ..
We are running an Asterisk 13 server which is having a strange problem, where on calls which are received from the PSTN and then forwarded out to the PSTN again there is no audio for the first 10 seconds of the call. At the 10 second mark audio sta..
Does anyone know of a way to have a call go to a particular context when a302 Moved is received in response to an invite? This is with chan_sip. We tried setting __TRANSFER_CONTEXT but it didnt seem to have any effect. Basically if a remote device retu..
We have a problem where one fax ATA connected to Asterisk works, and another ATA with the same model and firmware does not. Both are configured to use T38.Basically the call comes in to Asterisk which then does a Dial to the ATAthats registered via S..
I see some emails about a Dahdi compilation problem with linux/pci-aspm.h:No such file or directory two years ago, which suggest trying the nextbranch.Did this change go into a Dahdi release, and if so which version number(s)please?..
We have a commercial client who wants automatic fax detection, but the existing functionality in Asterisk doesnt quite meet their needs. Were willing to pay for a patch to do the following:1. Limit the automatic fax detection to the first X seconds..