Getting 488/Unacceptable Here on external phone calls

JS
Jason Stäuble
Thu, Dec 8, 2016 5:31 PM

Hi,

In my project i have a problem when making external calls.
By that I mean when i make a phone call to another phone in the office (with "phone-number" 7099) I can establish a call but when I make an external call (with phone-number 00791232429) I cannot  establish the call.

I also tried with several SIP Softphone applications like Zoiper and with them it worked.
I compared SIP Trace of both and could not find a lot of differences.

The SIP-Flow with pjsip console application looks like this:

ME  -->    DESTINATION : [SIP/SDP] INVITE
DESTINATION -->      ME : [SIP] 100/Trying
DESTINATION -->      ME : [SIP] 183/Session Progress
DESTINATION -->      ME : [SIP] 488/Not Acceptable Here
ME  -->    DESTINATION : [SIP] ACK

The SIP-Flow with Application like Zoiper looks like this:

ME  -->    DESTINATION : [SIP/SDP] INVITE
DESTINATION -->      ME : [SIP] 100/Trying
DESTINATION -->      ME : [SIP] 183/Session Progress
DESTINATION -->      ME : [SIP/SDP] 183/Session Progress
DESTINATION -->      ME : [SIP/SDP] 200/OK
ME  -->    DESTINATION : [SIP] ACK

However, internal calls are working fine with pjsip:
ME  -->    DESTINATION : [SIP/SDP] INVITE
DESTINATION -->      ME : [SIP] 100/Trying
DESTINATION -->      ME : [SIP/SDP] 200/OK
ME  -->    DESTINATION : [SIP] ACK

Both pjsip and working SIP softphone are using PCMU/8000 Codec, so i guess that should not be the problem?

Hi, In my project i have a problem when making external calls. By that I mean when i make a phone call to another phone in the office (with "phone-number" 7099) I can establish a call but when I make an external call (with phone-number 00791232429) I cannot establish the call. I also tried with several SIP Softphone applications like Zoiper and with them it worked. I compared SIP Trace of both and could not find a lot of differences. The SIP-Flow with pjsip console application looks like this: ME --> DESTINATION : [SIP/SDP] INVITE DESTINATION --> ME : [SIP] 100/Trying DESTINATION --> ME : [SIP] 183/Session Progress DESTINATION --> ME : [SIP] 488/Not Acceptable Here ME --> DESTINATION : [SIP] ACK The SIP-Flow with Application like Zoiper looks like this: ME --> DESTINATION : [SIP/SDP] INVITE DESTINATION --> ME : [SIP] 100/Trying DESTINATION --> ME : [SIP] 183/Session Progress DESTINATION --> ME : [SIP/SDP] 183/Session Progress DESTINATION --> ME : [SIP/SDP] 200/OK ME --> DESTINATION : [SIP] ACK However, internal calls are working fine with pjsip: ME --> DESTINATION : [SIP/SDP] INVITE DESTINATION --> ME : [SIP] 100/Trying DESTINATION --> ME : [SIP/SDP] 200/OK ME --> DESTINATION : [SIP] ACK Both pjsip and working SIP softphone are using PCMU/8000 Codec, so i guess that should not be the problem?