I start VOIP training in a couple of weeks. 8 hours all week. I have the criteria and want to get a jump on it. One was reading SIP Traces. This video was pretty easy to grasp. Thanks.
I work at GTT. I do internet troubleshooting and recently acquired VOIP. We have a couple of VOIP engineers flying up from Dallas to train us. 8 hour for 5 days next week =fried brain.
Thank you very much. Out of all the SIP videos on RU-vid yours was the most effective in explaining the SIP call flow. Can you do a video on building a SIP trunk. I don't understand the basics needed to build one (coming from a Avaya perspective)
Excellent video...you have explained things in much better simpler way than my teachers !!! I have a question to ask you-I have a site using SIP trunks. When I dial a 4 digit extension from a shared line, it gives my fast busy tone ONLY in SRST mode If I dial the same number from same phone in normal mode, all works well. My suspicion is Cisco Toll Fraud Prevention is blocking my calls...maybe,,,, For this, I have captured traffic using wireshark (when I unable to dial the number) but I don't know where in wireshark in will give an idea that it is indeed toll fraud prevention blocking calls!!!I ave been looking and looking. Any suggestions?
I would look for your specific call in the trace and look to see why it was rejected. If Cisco uses SIP protocol for their stations, you should at least see the call being initiated. Once you see that, follow the SIP flow to see what the rejection code was. It may not tell you specifically, but it may give you a better idea.
Terrell,, Thank you for making the videos and sharing what you have learned with those that are learning it...Such is the circle of life. Subscribed and looking forward to more from you.
I would appreciate your giving a presentation on how to set up wireshark or some other free app to enable us to see what you are seeing. Thank you for taking your valuable time to help us
For making an external call from my PBX to outside by means of sip trunk , i have experienced an problem. The calls are diconnected after one mins. The sip trunk were configured in SBC. I donot know what should i do? I checked the wireshark trace and found that the Release Message were coming from the Phone . Internal calls are woking fine. Will be glad to have your suggestion. Thanks in advance.
Hi Terrell. This is a good video. I was wondering like Crusty Tackleford,(who asked 10 months ago) how you setup your equipment to be able to capture these packets with Wireshark. Please shade light on this
Hello, not sure if you still respond here. Just trying. I watched your outbound tutorial on SIP Troubleshooting for Beginners - Outgoing Call Trace Review. I am needing to understand how to set up traces to show the RTP stack. Currently I am unable to figure out how to incorporate the RTP/Audio information in my PCAPS. Any help on this would be appreciated. Thanks, Raheem
Can you please show how and where you setup the trace in relation to where your equipment and firewall is. If you could maybe throw in a diagram that would be great. I'm confused where you took the trace and what port was setup for mirroring and such.
Wow, this is definitely the video I was looking for, really good explanation, Terrel I do not know how to express how grateful I am with this video, I hope you can make more videos such as this one... Thanks :) Subscribed!!!!
how would I answer this questions? You have observed the INVITE - 200 OK - ACK three-way handshake during the call setup. What messages are exchanged for tearing down a call session?
Hi Terrell, this is very helpful. I am supposed to demonstrate the same to my colleagues in a training session so that I can easily explain about encrypted SIP in Microsoft Lync calls. I searched for a completed SIP call capture file in wireshark site, but couldn't find one that is as good as this. I would be glad if you could share this capture file with me. Thank you.
hello sir what does it mean 401 Unauthorized and 500 Internal server error . SIP outgoing call not working after analyses the packet flow i received this
from where can i get the sip training in Riyadh Saudi Arabia, a physical taring as i am working in STC and can understand the system, traces, putty etc , please your feedback
Hi Ayman, It will be in the messaging and is determined during the call setup based on my understanding. All media gateways should detect DTMF tones, they let the PBX/CFS/Switch or whatever you're using know it detected tones and what the digits were so the PBX etc can determine if they have a feature associated with it or not (flashook/3way calling etc). If it doesn't those digits get passed over to the endpoint device of the sip trunk the call routed over; However, as I stated earlier whether the DTMF tones are passed inband or out-of-band is determined when the call is first setup based on my experience so you should see it in your standard sip messaging capture.
Great video Terrell. I'd love to setup an ADTRAN 908e in a lab in order to capture a pcap file with a good working test calls. Thanks to your video's I think I have handle on setting up for the tcpdump. I'm just not sure how to setup an ADTRAN 908e for a lab environment. Do you have any suggestions? I've contacted ATRAN, but no luck yet. The ADTRAN does have some great debug commands that I've used many times, but not as good as what you'll get in Wireshark.
Hello, @Terrell. Great Video btw. Is there any chance to receive RTP packages ( from freeshwitch) while outgoing / incoming calls only (no active session)
