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Asterisk skips first DTMF
Asterisk Incoming Calls Don't Route to ExtensionsSkype Connect as SIP/Trunk for AsteriskHow to configure Asterisk to send audio before call is establishedAsterisk: Authenticated SIP trunkcant make outbound calls - asteriskAsterisk ignoring sip_domain part of URIAsterisk inbound trunk unauthorised without allowguest=yesAsterisk: Join 2 inbound callsAsterisk and FXO issue over DDNSCan't dial through SIP trunk: FreePBX/Asterisk
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I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I complete a call and send DTMFs, Asterisk Server always ignores the first dtmf i send, it answers a 200 OK to the endpoint but do not forward the signal to the other call leg. From the second DTMF on, it answers 200 OK and forward the SIP INFO to the other leg normaly. Have you guys ever seen this? I did the same deploy on a lab environmet and got the same results.
asterisk sip
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I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I complete a call and send DTMFs, Asterisk Server always ignores the first dtmf i send, it answers a 200 OK to the endpoint but do not forward the signal to the other call leg. From the second DTMF on, it answers 200 OK and forward the SIP INFO to the other leg normaly. Have you guys ever seen this? I did the same deploy on a lab environmet and got the same results.
asterisk sip
add a comment |
I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I complete a call and send DTMFs, Asterisk Server always ignores the first dtmf i send, it answers a 200 OK to the endpoint but do not forward the signal to the other call leg. From the second DTMF on, it answers 200 OK and forward the SIP INFO to the other leg normaly. Have you guys ever seen this? I did the same deploy on a lab environmet and got the same results.
asterisk sip
I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I complete a call and send DTMFs, Asterisk Server always ignores the first dtmf i send, it answers a 200 OK to the endpoint but do not forward the signal to the other call leg. From the second DTMF on, it answers 200 OK and forward the SIP INFO to the other leg normaly. Have you guys ever seen this? I did the same deploy on a lab environmet and got the same results.
asterisk sip
asterisk sip
asked Jun 10 at 10:48
Roberto NevesRoberto Neves
112 bronze badges
112 bronze badges
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don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.
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1 Answer
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active
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don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.
add a comment |
don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.
add a comment |
don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.
don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.
answered Jun 12 at 15:34
Roberto NevesRoberto Neves
112 bronze badges
112 bronze badges
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