Tag : SIP Provider
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asterisk users,I have a strange behaviour with asterisk and error code forwarding in asterisk 11.Please find below my setup:Phone -> ASTERISK -> SIP TRUNK PROVIDERA phone start a call, asterisk start a leg to my SIP trunk provider. I have a simple dialp..
Think janitorial or security kind of thing where an employee goes from location to location.Theyre supposed to clock in when they get to a site using a phone at that site to prove the..
my SIP provider requires 10 digits for all outgoing calls; Users dial 7digits for outgoing.Here is how I added the area code to all outgoing calls in Asterisk 1.8 Extensions.conf; Adding Area code and striping 7 for local numbersexten => _7XXXXXXX,n,Set(CALLERID(all)..
Im struggling to separate inbound calls fro a SIP provider that does not send DID. I have tried…….sip.com/12345678 on register string different context=from-no-did Port not possible as only support 5060Anysuggestions?Th..
Im working with a SIP provider to try and transition our sip connection with them to PJSIP.I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages.Im currently running Aster..
,I have the following situation:Local T.38 endpointASTERISKSIP provider (with T.38 support)I am trying to send a fax from my local T.38 endpoint to arbitrary external fax numbers (which I am not in control of, so I dont know if the other end suppo..
Greetings- Working with the T.38 gateway functionality that is sparsely documented [1] , Im attempting to get the following functional: Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in question) -> SIP Provider The problem ..
,I have a fresh install of Asterisk 12.0.0 and Im going to use it only as a client. Im trying to SIP REGISTER with a remote SIP provider.The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of ..
My target system is :PSTNSip ProviderRouter with fw/NATAsteriskSIP PhonesAsterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I dont have any problem (yet) with either casual incoming or outgoing calls.To work aro..