Basic Sip Quesiton


What am I missing please? sip show registry shows that I am registered.

[general] register => ; ; [] bindportP60 ;you can use different port if the default is blocked bindaddr=00.0.0 ;binds to all

;this is for codec negotiation between the useragent and asterisk disallow=al allow=ulaw allow=aaw allow=g729 allow=gsm

context=incoming ;default context where incoming calls are passed. this should be the context where your local user.s extensions reside

[outbound-trunk] ;this is the second section of you sip.conf file. Here you can create your trunk through which you will throw your outgoing calls to axvoice. type=peer dtmfmode=rfc2833 canreinvite=yes

Asterisk Users 3.3 years ago 2 Answers

Asterisk 1.8.13 PlayTones App


Hi, i'm trying to implement a Playtones App into my IVR. If a invalid numer is entered, Asterisk should play the info tone defined at the indications.conf. My Extensions.conf exten => i,1,Set(CHANNEL(language)=de) exten => i,2,Progress() exten => i,3,PlayTones(info) exten => i,n,Playback(fettefinger) exten => i,n,Wait(2) exten => i,n,StopPlayTones() exten => i,n,Goto(i,3) The Console shows Invalid extension '5' in context 'support' on SIP/200-0000000a == CDR updated on SIP/200-0000000a -- Executing [i@support:1] Set("SIP/200-0000000a", "CHANNEL(language)=de") in new stack -- Executing [i@support:2] PlayTones("SIP/200-0000000a", "info") in new stack -- Executing [i@support:3] Playback("SIP/200-0000000a", "fettefinger") in new stack -- <SIP/200-0000000a> Playing 'fettefinger.alaw' (language 'de') -- Executing [i@support:4] Wait("SIP/200-0000000a",…

Asterisk Users 3.3 years ago 0 Answers

Queue Member Login From IAX Trunk



i got two Asterisk servers connected with IAX2.

At server 1 i hosted a queue named support. Now i want the Agent to login from SIP Phones Connected to server B. Therefore i wrote a extension like

exten => 105,1,Authenticate(1234) exten => 105,n,AddQueueMember(support,,) exten => 105,n,Read(AGENT_SIP,agent-newlocation) exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})}) exten => 105,n,Playback(agent-loginok) exten => 105,n,Playback(vm-goodbye) exten => 105,n,Hangup

Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the entered number in agent-newlocation. Do you have any ideas how to solve that?

Best regards Jakob

Asterisk Users 3.3 years ago 3 Answers