well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate.
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Let me know if you need more information. Thanks, Vivek On 11/11/07, Ryan Newington <[EMAIL PROTECTED]> wrote: > > > > Hi Vivek, > > > > I'm not sure what you mean, could you explain further? > > > > Regards > > > > Ryan > > > > > > *From:* [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] *On Behalf Of *Vivek Shrivastava > *Sent:* Monday, 12 November 2007 1:21 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] RTP traffic not being forwarded > > > > Hi Ryan, > > > > I was just wondering if they need to be according rtp.conf. ( or you may > need to modify rtp.conf) > > > > Regards, > > > > Vivek > > > > On 11/11/07, *Ryan Newington* <[EMAIL PROTECTED]> wrote: > > Hi Vivek, > > > > The SIP port is set to the standard port 5060. The RTP ports as far as I > know are random ephemeral ports between 63000 and 64000. > > I can change the port range on the media server, asterisk and the device, > but neither seems to help. > > > > My diagram below is probably misleading. The RTP traffic flow that I see > is as follows (one way traffic into Asterisk) > > > > SIP Phone <---> Media Gateway *--->* Asterisk *<---* SIP Phone > > > > Ryan > > > > > > *From:* [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] *On Behalf Of *Vivek Shrivastava > *Sent:* Sunday, 11 November 2007 5:19 PM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] RTP traffic not being forwarded > > > > Hi Ryan, > > > > Are the SIP and RTP ports are randomly selected or there are specific > ports for these? Unchecking > > random port selection option on the device/softphone may help. > > > > --Vivek > > > > On 11/10/07, *Ryan Newington* <[EMAIL PROTECTED]> wrote: > > Hi Luki, > > Thanks for your advice. I've checked the firewall and it is set to allow > all incoming traffic. I changed the media port range as well with no > success. > > Some calls work fine. This is the configuration that doesn't work. The RTP > traffic passes along the chain fine, but the Asterisk server doesn't do > anything with the packets it gets from the near-end SIP phone and the media > gateway. > > SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone > > An asterisk internal call will work fine. Eg; > > SIP Phone <-> Asterisk <-> SIP Phone > > Regards > > Ryan > > > > -----Original Message----- > From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] > ] On Behalf Of Luki > Sent: Sunday, 11 November 2007 12:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] RTP traffic not being forwarded > > > When using 'rtp debug' on the asterisk console, it shows that it is > > receiving traffic from one endpoint, but not the other. A wireshark > trace > > reveals it is actually receiving traffic from both ends. > > Sounds like a firewall issue. Wireshark shows what's "on the wire", > i.e. before iptables. The packets are being dropped for whatever > reason and never reach the asterisk process. Check your iptables and > RTP port range, and perhaps try changing it. > > Luki > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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