Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-30 Thread Philip Prindeville
Andres wrote: I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-29 Thread Philip Prindeville
Philip Prindeville wrote: Well, things just got a lot more interesting... Adding Monitor() to an extension ends the one-way voice problem on inbound calls! So an incoming call gets handled as: [ctc-incoming] exten = 208345,1,Noop() exten = 208345,n,Log(NOTICE: RDNIS:

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-28 Thread Sergio
A similar issue happens to us. Make sure that, for inbound AND outbound calls rtp packets are reaching the other endpoint. If a NAT device(s) is between the endpoints make sure that the device NATs the traffic on BOTH ways (inbound AND outbound). Regards On Saturday 27 September 2008 23:54:37

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-28 Thread Kristian Kielhofner
On Sat, Sep 27, 2008 at 5:54 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-28 Thread Philip Prindeville
Well, things just got a lot more interesting... Adding Monitor() to an extension ends the one-way voice problem on inbound calls! So an incoming call gets handled as: [ctc-incoming] exten = 208345,1,Noop() exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI: ${CALLERID(ani)})

Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-28 Thread Andres
I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to

[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

2008-09-27 Thread Philip Prindeville
I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way