[Asterisk-Users] SIP - SIP Problems

[Tim Pushor]

Ian

I don't run X on any of my servers. I always pre-capture the data with 
tcpdump to analyze with a windows or linux + X system running ethereal.

tcpdump -s 1500 -w file.out -i <int> <filter expression>

Will start tcpdump and write packets matching filter_expression to 
file.out. Press ctrl-c after you have captured what you want, and 
transfer this file to the system that you run ethereal on, and load it 
(assuming max M[TR]U of 1500).

Ian Pattison wrote:

<snip>

>Packet decodes are my next step... has anyone here ever successfully had Ethereal running in text-mode only? My * box does not have X installed and is only accessible via SSH.
>
>Thanks,
>
>Ian
>
>  
>
>>>>"Rod Bacon" <rod.bacon at empoweredcomms.com.au> 06/04/2005 20:59 >>>
>>>>        
>>>>
>I'd personally be using Ethereal to look inside the SIP messages for the SDP 
>info and checking the source/destination of the resultant RTP stream. 
>One-way audio is typical of NAT issues. Although you are running a VPN (of 
>sorts) I suspect that your SDP messages are getting screwed up somewhere.
>
>What are the asterisk NAT settings in effect for each of the SIP phones? I'd 
>be inclined to turn them both ON to ensure that symmetrical RTP in being 
>used. Also make sure that canreinvite is OFF for both.
>
>
>----- Original Message ----- 
>From: "Ian Pattison" <ianp at technologyassociates.ca>
>To: <asterisk-users at lists.digium.com>
>Sent: Thursday, April 07, 2005 4:49 AM
>Subject: [Asterisk-Users] SIP - SIP Problems
>
>
>Hi Everybody...
>
>Continuing the litany of problems I'm experiencing with my new system I'm 
>=etting issues connecting between SIP phones.
>
>A bit of background... I have an asterisk server running in a central 
>=ocation where I have two incoming analog lines connected to FXO ports, =wo 
>analog phones connecting to FXS ports and a single SIP phone. In =ddition I 
>have a remote site connected via a CIPE VPN (ok..ok I know it's =ot a real 
>VPN...) with a single SIP phone.
>
>Calls initiated from the remote SIP phone to the central SIP phone often 
>=ave trouble... the user of the central phone cannot hear anything from =he 
>remote phone although everything is heard at the remote phone. If the =emote 
>phone calls either outside or to one of the Zap phones there is no =rouble. 
>If the local SIP phone calls the remote SIP phone there is no =rouble. Both 
>phones are from the same vendor, have the same firmware and =he same 
>configuration with the exception of phone number, PIN, IP address =tc.
>
>What am I doing wrong here?
>
>Ian
>
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