[asterisk-users] Re: Question about DSP in Digium card
well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine Those are the reasons we choose IAX as acess protocol to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, .. Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about DSP in Digium card
Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use a Half E1 with Asterisk?
Hello I want to know which hardware I have to use in order to use a half E1 with Asterisk (the second half will be used by a PABX PANASONIC). I have already a succesfull experience in Asterisk with an entire E1 (TE110P card) or 4 analogic channels (TDM400P) but I have no idea how physically connect to a half E1 Thank you for your help _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line detection with TDM400P
Hello I just want to have a confirmation, line status detection (with digium TDM400P) is highly not reliable outside of US. With busydetect=yes and callprogress=yes I can experience very strange phenomenons (randomilally occurs) like pick up not detected or hang up not detected. I'm in Israel , somebody knows how to improve this detection (it's very important for me to get call status live). Thank you _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Way Audio....in the middle of a call
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock [EMAIL PROTECTED] wrote: I experienced this today.Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times.Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume.PhilFrederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto: [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
