2020. 3. 6. 00:31ㆍ카테고리 없음
Hi all,I have been trying to make SIP calls from my Cisco 5300 RTR terminate to my ISP ( TATA) but still can’t get it to work call are getting disconnected after first Ring. Below are the equipment Involve.E0 SIP TrunkNortel Switch - cisco 5300 RTR - ISPI see all calls hitting the dial-peer 4007 pots, and then hitting the dial-peer voice 4003 voip (test number is 34). When I do a SIP TRACE using “debug ccsip messages” I see that my ISP is requesting “Require: timer”, like if the CALL setup Timer Expired. I increase the timer setup under “ Sip” and set it to “ min-se 48000” but still calls are being disconnected. I already checked my ISP and they are saying that the call is being disconnected on my side, and that they are using “codec g729r8”. I have the codec set to G729r8 on my side. From this logs now we see that the issue is with your provider.We get a 180 ringing at 23:59:52.Mar 26 23:59:52.431: Received:SIP/2.0 180 RingingThe gateway didnt get a 200 OK and after 20 sec the call was terminatedMar 27 00:00:12.435: Sent:CANCEL sip:34@66.198.40.9:5060 SIP/2.0You need to go back to your ITSP and find out whats going on.Why are they not sending a 200 OK response to our INVITE.Please rate all useful posts'opportunity is a haughty goddess who waste no time with those who are unprepared'.
Danny,Yes its yoiur gateway that is sending the BYE immediately after receiving the 200 OK from the provider. It looks like your gateway expects some SDP in the 200 okay, but your provider is already sending SDP in 180 ringing.You can try and disable PRACK and lets see if your ITSP will send media in 200 OK. According to the RFC the only time when the SDP attributes in 180/183 response differ to 200 OK is only when you use PRACK. SO if you disable PRACK and your ITSP doesnt send SDP in 200 OK exactly as is in 180 ringing, then theier implementation is illegal.Please rate all useful posts'opportunity is a haughty goddess who waste no time with those who are unprepared'. Hi Aokanlawon,I disable the ' PRACK' using ( no rel1xx require '100rel') and made a TEST calls but still the calls are Disconencted.
From this logs now we see that the issue is with your provider.We get a 180 ringing at 23:59:52.Mar 26 23:59:52.431: Received:SIP/2.0 180 RingingThe gateway didnt get a 200 OK and after 20 sec the call was terminatedMar 27 00:00:12.435: Sent:CANCEL sip:34@66.198.40.9:5060 SIP/2.0You need to go back to your ITSP and find out whats going on.Why are they not sending a 200 OK response to our INVITE.Please rate all useful posts'opportunity is a haughty goddess who waste no time with those who are unprepared'. Hi Aokanlawon,Finally the calls are Terminating succesfully.
Sip Invite Format
It had to do with the 'PRACK' as you mensioned earlier. I called the ITSP at fisrt they were requesting the 'PRACK' but then they disable that option on thier end,and they Never Informed me, I went and disabled the Option on my side and the calls Terminating Succesfully, I think the ITSP did more changes on thier side becasue I have the same configuration, I was Using when I started the TEST.You Help me ALOT in Identifying the Problem.thanks for evreything,Wish you the BEST.take care.Danny. Hi Aokanlawon,Yes, calls are working without 'PRACK'Below is the configuration I used but I am sure the ITSP did some changes on thier side, becuase this is the same Configuration I was using when we started the SIP Trunk connectionit never work and then I ADD the 'PRACK' which made calls termiante but get disconnected, and finaly at the END I had to remove the 'PRACK' which Made the calls terminate Succesfully,LACK of COMMUNICATION with the ITSP.LOL. And THANKS AGAIN for YOUR HELP.Below is the COnfig I Am using,!voice service voipfax protocol pass-through g711alawh323no ras brqh245 caps mode restrictedsipbind all source-interface FastEthernet0!!dial-peer voice 4003 voipdescription OUTTATASONUSpreference 1destination-pattern 34session protocol sipv2session target ipv4:66.198.40.9:5060dtmf-relay rtp-nte h245-alphanumericno vad!