I tired to get inbound to work without having to manually edit any of the.conf files. All of these changes I made were used through the GUI and not editing any of the.conf files individually. I will post my trunk setting to show what I have gotten to work. Trunk name: EXXXxxxXXXX01 username=EXXXxxxXXXX01 type=friend secret=XXXXXxxxxxXXXXXxxxxx. Check 'in Service' box 3) Setup SIP URI - ADD a channel and set this to - Make sure to set the groups to something unique. I used group 420 for incoming and outgoing.
Active1 year, 1 month ago
I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration:
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However, whenever I try to place an outgoing call (through the same trunk) I have a 'all lines busy' signal from asterisk. If I enable SIP DEBUG this is what I get (apparently my call is being rejected due to an invalid alias at the other side, which I can't control since it's my VOIP provider):
Any ideas of what might be wrong on my side of things?
If I connect a simple softphone to my VOIP provider, it works flawlessly (incoming and outgoing calls).
Pablo Santa CruzPablo Santa Cruz
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2 Answers
My guess is that your Caller ID is offending them. Are you setting to anything other than your actual assigned DID?
Based on:
MichelV69MichelV69
This is a know bug from Asterisk when you use the non standard sip port 5060 for your server. The bug is discussed here https://issues.asterisk.org/jira/browse/ASTERISK-24767 .
You should be able to correct this using fromdomain=prepago.com.py:5060 but Asterisk ignores the directive port and rewrite the from as From: 'sip:[email protected]:5061'.You can patch Asterisk code and recompile it or use the standard sip port in your server.
Victor Gonzalez ChamorroVictor Gonzalez Chamorro
Trix Box Installation GuideNot the answer you're looking for? Browse other questions tagged asteriskvoiptrunkfreepbx or ask your own question.
Active6 years, 1 month ago
I am totally new to this PBX, I really don't know what is it and I wanna learn it ASAP, I've created an SIP peer and an SIP user, in Extensions.conf I need the code to route all outbound calls to that SIP Peer (which is a gsm gateway of GoIP). Here're my files
Extensions.conf :
User Configuration in Sip_additional.conf:
Peer Configuration in SIP.conf:
Please note that both my GSM Gateway and My SIP client are connected to the asterisk pbx successfully, The thing I request now is that whenever I dial a number from sip client, it should be dialed from the gsm gateway, if you need any kind of additional info please post a comment and you'll get a reply soon
user2466636
2 Answers
You need context like:
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But i highly recomend you read some book for beginners like 'asterisk the future of telephony'
arheopsarheops
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Guys This worked for me:
[from-internal] exten => _X.,1,Dial(SIP/gsm1/${EXTEN})
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user2466636
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