Simon, Thank you!! I used that psipdump, and it worked like a charm, it didn't take me long to get a call with no audio, my problem is now solved.
For everyone else on the list, here is my setup, and the solution to the problem: I have a SIP trunk with Unlimitel. I'm running Asterisk 1.2 on Fedora Core 4. I have a Cisco 837 router as my ADSL router + NAT + statefull firewall. My firewall was configured to allow all traffic from the Unlimitel SIP gateway address. For some reason, when a SIP call was being made from my asterisk box to Unlimitel, the RTP session would SOMETIMES connect to a different IP address then the SIP session. Thanks to psidump and ethereal I was able to see this difference. The solution for me was to simply to open my firewall to all of Unlimitels IP addresses that they use for SIP and RTP. On Thu, 2007-18-01 at 17:03 -0500, Simon P. Ditner wrote: > It all sounds suspiciously like NAT/Firewall traversal issues. Perhaps > double-check that canreinvite=no is set in your sip.conf for your phones? > > This likely falls into the "extreme" category of troubleshooting > techniques, but my advice is to go grab this application, and leave it > running on your asterisk servers (with an appropriate script to purge old > data), and gateways if possible: > > http://sourceforge.net/projects/psipdump/ > > It's a small program that captures the SIP and RTP data for calls and > saves it into unique caller-callee-timestamp-sessionid files in pcap > format. > > So if you can get people to report the time a problem occurred +- a few > minutes, you should be able to pick out the right capture file, load 'em > up in ethereal, and see what SIP messages are flying around & where the > RTP traffic thinks it's supposed to be going. > > re, > spd > > On Thu, 18 Jan 2007, Mark Rzepa wrote: > > > I'm having a similar problem. I have asterisk transferring calls to my > > cell phone if I don't pick up my extension, it works most of the time, > > however sometimes no audio comes out from my cell back to the caller. > > I've been trying to find a way to determine if it's me or my service > > provider, but since all calls that I make/receive with my SIP phone work > > 100% of the time, I think it's asterisk. > > > > Does anyone have an idea on how to debug this? > > > > To make troubleshooting harder, I can't replicate the problem on demand, > > sometimes it works, sometimes it doesn't. > > > > On Thu, 2007-18-01 at 16:34 -0500, Henry.L.Coleman wrote: > > > It may be worth checking the cable(s) > > > Henry L.Coleman CEO > > > *VoIP-PBX* 1-866-415-5355 > > > Toronto Ontario > > > Canada > > > > > > > > > > John, > > > > > > > > The fact that the audio problems are solved by transferring from one > > > > phone to another would indicate to me that it's between the phone, and > > > > the Asterisk server. > > > > I take it that it's a direct connection through one switch from the > > > > phones to the server? > > > > > > > > I have a client that received fairly heavy call volume, a few hundred > > > > calls a day, on IP501's using Static Queues on 1.2.13, and 1.2.14 (as of > > > > 2 weeks ago). They have experienced no similar problems, so I can't say > > > > "me too". > > > > > > > > What firmware version are these phones running? We use 2.0.1 with no > > > > problems. > > > > > > > > Can you get a packet dump from the server and see if it's sending the > > > > phone data? > > > > > > > > Chad > > > > > > > > > > > > > > > > -----Original Message----- > > > > From: John Van Ostrand [mailto:[EMAIL PROTECTED] > > > > Sent: January 17, 2007 6:41 PM > > > > To: [email protected] > > > > Subject: [on-asterisk] No audio on some calls > > > > > > > > This is one that we've been fighting for a while. > > > > > > > > We have a small install (8 extensions) using [EMAIL PROTECTED] with > > > > Polycom 501 SIP > > > > phones. Every once and a while one side of an incoming call's audio is > > > > missing. The callers can hear the employee but the employee can't hear > > > > the caller. They have been able to handle it in the past by transferring > > > > the call to another extension. > > > > > > > > To make this worse, it seems to only be occurring with two phones. We've > > > > replaced the phones three times and the report from the customer is that > > > > it fixed it, then a week or so later we hear that it's back. > > > > > > > > More oddities, is now they say they've had it occur on an outgoing call > > > > too. > > > > > > > > The two phones that it is occurring are in a static queue and most of > > > > the calls they will receive are from the queue. > > > > > > > > The reports from this customer are pretty sketchy, but they have been > > > > insistent about the audio problems. > > > > > > > > Any ideas. > > > > > > > > -- > > > > John Van Ostrand Net Direct Inc. > > > > CTO, co-CEO 564 Weber St. N. Unit 12 > > > > Waterloo, ON N2L 5C6 > > > > [EMAIL PROTECTED] ph: 518-883-1172 x5102 > > > > Linux Solutions / IBM Hardware fx: 519-883-8533 > > > > > > > > > > > > --------------------------------------------------------------------- > > > > To unsubscribe, e-mail: [EMAIL PROTECTED] For additional > > > > commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > > > > > > > > --------------------------------------------------------------------- > > > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > > > > > > > > > > --------------------------------------------------------------------- > > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] >
