Another nice sip packet is sngrep Shows realtime the sip flows But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. ?????? 6 ????? 2016 10:05 PM,? "Steve Edwards" <asterisk.org at sedwards.com> ???:> On Wed, 6 Jul 2016, Victor Villarreal wrote: > > If you experience problems with inbound calls from a SIP trunk or >> provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set >> debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from >> where it is supposed to come call. >> > > Another path to enlightenment is to use tcpdump to capture the packets to > a file and then use wireshark. > > Wireshark has a 'Telephony' menu and a 'SIP Flows' menu item where it will > list all of the SIP packets. You can sort by column to help locate the > packet of interest. > > Once found, you can click on 'Flow Sequence' and it will pop up a window > showing the 'dialog ladder' that includes that packet. As you click on each > packet in the flow, the main wireshark window will re-position to that > packet so you can examine it in detail. > > Also on the 'SIP Flows' window is a 'Play Streams' button. Kind of scary > how easy this is... > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/c8599f54/attachment.html>
Hi, I am using pcapsipdump for debugging sip calls. when I have to debug a call, pcapsipdump generates two files per call... one for the sip dialog between the client (softphone) and the server (asterisk) and one for the sip dialog between the server (asterisk) and the sip registrar... is there a way to get this into one file ? the objective is to see both sides of the call in a single ladder diagram or just to have more comfort in analyzing the full flow within wireshark. If this is not possible, is there a free tool for sip (together with rtp) debugging that is able to catch the full sip flow between both ends of one call in a single file (per call) with pcap compatibility (including the rtp packets)? thank you yves
Jean Aunis
2017-Jan-17 11:34 UTC
[asterisk-users] pcapsipdump or general sip debug question
Hello, There is a built-in tool in Wireshark for this : menu Telephony => Voip Calls, the select your call and click on "Flow Sequence". Best regards Jean Aunis Le 17/01/2017 ? 12:27, Yves a ?crit :> Hi, > > I am using pcapsipdump for debugging sip calls. > > when I have to debug a call, pcapsipdump generates two files per > call... one for the sip dialog between the client (softphone) and the > server (asterisk) and one > for the sip dialog between the server (asterisk) and the sip > registrar... is there a way to get this into one file ? the objective > is to see both sides of the call in > a single ladder diagram or just to have more comfort in analyzing the > full flow within wireshark. > > If this is not possible, is there a free tool for sip (together with > rtp) debugging that is able to catch the full sip flow between both > ends of one call in a single file > (per call) with pcap compatibility (including the rtp packets)? > > thank you > yves > >