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I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

...o using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn?t receive sound... Andre Lomonaco -----Mensagem original----- De: Low, Adam [mailto:ALow@Prioritytelecom.com] Enviada em: Friday, September 26, 2003 9:06 AM Para: 'asterisk-users@lists.digium.com' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will nee...

All, seems I too am suffering from posts to the list and being accused of SPAMing .... -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: 26 September 2003 20:48 To: alow@prioritytelecom.com Subject: RE:RE: [Asterisk-Users] RTP routing.. <http://antispam.uol.com.br> <http://mail.i.uol.com.br/tirateima_txt.gif> <http://www.uol.com.br> Ol?, Voc? enviou uma mensagem para andersoncbr@uol.com.br Para que sua mensagem seja encaminhada, por favor, <...

All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the

Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun

All, Has anyone had any thoughts/discussion on providing a malicious call trace feature within Asterisk. Most legacy PBX's support this feature which allows a handset user to indicate using DTMF during a call that it's a malicious call which instructs the PBX to send a specific Q931 message over the ISDN to the providers switch telling it to log the call details as malicious for later

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

...t; All, seems I too am suffering from posts to the list and > being accused > > of SPAMing .... > > > > > > -----Original Message----- > > *From:* AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] > > *Sent:* 26 September 2003 20:48 > > *To:* alow@prioritytelecom.com > > *Subject:* RE:RE: [Asterisk-Users] RTP routing.. > > > > <http://antispam.uol.com.br> <http://www.uol.com.br> > > > > Ol?, > > > > Voc? enviou uma mensagem para *andersoncbr@uol.com.br* > > Para que sua mensagem seja encamin...

Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

...I board in the PC of which I don't have. Any experiences/comments most appreciated. Rgds, Adam _____________________________________________________________ Adam J. Low Tel: +31 20 778 2740 Senior Network Architect Fax: +31 20 778 2600 Priority Telecom Corporate Email: alow@prioritytelecom.com ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any...

Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

...00P in Germany? (I assume I have to exchange some > wires.) > It would be operated inside a internal PTSN network, so my question > is mainly technical, not about permission. > > > Roger. > > > > --__--__-- > > Message: 9 > From: "Low, Adam" <ALow@Prioritytelecom.com> > To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Need Europian vendor for Digium hardware. > Date: Mon, 26 Jan 2004 10:58:03 +0100 > Reply-To: asterisk-users@lists.digium.com > > http://www....

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

Hi All, I've had a great success these last couple of days on bringing up Asterisk with a load of Cisco 7940 phones and a couple of soft phones as well. I'd really like to try and link this in to my companies PBX so I can call in from the PSTN as well. I understand there is a charge for the module that would allow me to use my old Dialogic board, can anyway provide some more information

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

I too have been having SIP problems the last couple of days, I get the same message as Louis-David but in this setup only: PSTN >-ISDN-> AS5300 >-SIP-> * I can make outbound calls no problem but inbound calls seem to stall, according to 'sip debug' it just says 'Ignoring this request' but I cant establish why .... > -----Original Message----- > From: William

Hi All, I've been developing all sorts of applications for use on our 79xx handsets but am having great difficulty with formatting, I just can't seem to be able to produce a line feed between lines on the stuff actually displayed on the phone. Has anyone else has experience or success with this ? Cheers, Adam ********* DISCLAIMER ********* This message and any attachment are

I am guessing you are running without reinvite's, I'm running with reinvite's with latest CVS release and 79x0 phones without any issues with conferencing... > -----Original Message----- > From: Adam Rothschild [mailto:asr@latency.net] > Sent: 08 October 2003 15:49 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 phone and >