Tag : sip
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AllAfter a Dial() I get:WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type SIP (cause 20 – Subscriber absent)if the subscriber is not registered.Is there a way from dialplan to know, *before* Dial(), if a destinat..
I have a problem where SIP calls from some providers are dropping at 15 minutes.The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server.Below,Client is the IP address of the clients host (running FPBX-2.8.1(1.8.20.0)OpenS..
I am having a small problem that is driving me nuts.I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I w..
Group,I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue.Please help me.I have tried calling with two SIP end point forwarding , even that is not working,My dial plan l..
I am trying to set add a SIP Header to a call before adding it to the Queue.The dial plan sends the call to my macro to perform the work.When I use chan_sip, everything works as expected.When I use PSJIP support, its not adding the SIP header. Look..
I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset.I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here.From what I read this is usually codec related but b..
AllNoticed in sip.conf that the asterisk (v11) is sensitive to the order..
we have an issue where after a couple of days, a few random phones will lose registration. I dont notice any particular pattern. Out of 200, only about 5-10 will be suffering at any given time and we wont know until the user complains they are not receiv..
Running Asterisk 11 on CentOS 6.x using the DAHDI module with 8x PSTN analog phone lines for outside connectivity. Internally, I am using several models of Yealink SIP phones (e.g SIP-T32G) on a dedicated VoIP network, 192.1..
every bodyi have problem in receiving fax from e1 lines. this is my scenario:faxphone—-ericson pbx —e1—-asterisk—-sip—–zoiper-softphonewhen i send fax from zoiper, i can receive it successfully on the faxphone but when i send fax from faxpho..