similar to: SIP + NAT question

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Hi, On my linksys/sipura phones/ATA, there is a setting called "NAT Mapping Enable" and another called "NAT Keep Alive Enable" These settings must be on in my setup so that my phones/ATA remain connected to my * server. My setup is: Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 NAT, Static public IP) - Asterisk server. I was wondering:

I am having some oddness with the 11/11/2003 CVS of *. Specifically, outgoing audio to NuFone doesn't seem to be transmitted (I can hear the other side just fine). My firewall is set to allow all outgoing traffic, and the IAX2 connection is definitely established correctly. Also, I can watch UDP traffic going by on the firewall so I know that * is transmitting. This happens with X-Ten on

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

How are people handling call transfer with SNOM phones? We are okay with the "#" transfer workaround, but I worry about how that will work with other systems that expect me to be able to "press # to return to the previous menu" or similar. Thanks, --Ernest

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Good day all I cant get my grandstream bt-100 to register My asterisk is on a public ip and the phone behind a nat firewall I added nat=yes in sip.conf and did this on my grandstream set the GS to "SIP server=asterisk.yourhost.com" and leave "Outbound Proxy" empty * set the GS to SIP port 5060 and RTP port 5004 (and "Use random port=No") * set the GS to

I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm

Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Message: 11 Date: Mon, 29 Sep 2003 12:45:40 -0700 To: asterisk-users@lists.digium.com From: "Ernest W. Lessenger" <ernest@oacys.com> Subject: Re: [Asterisk-Users] Is somthing broken? Reply-To: asterisk-users@lists.digium.com At 12:33 PM 9/29/2003, you

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

InDATA <-read.table("C:/Data/May 2004/season.txt",header=T) X <- decompose(InDATA) print(X) Period Connections Q1 67519 Q2 69713 Q3 68920 Q4 69452 Q1 70015 Q2 59273 Q3 57063 Q4 65596 Q1 73527 Q2 58586 Q3 69522 Q4 60091 Q1 51686 Q2 63490 Q3 55702 Q4 53200 Q1 51033 Q2 48175 Q3 52709 Q4 50106 Q1 50855 Q2 43466 Q3 48190 Q4 41702 Q1 48747 Q2 51441 Q3 42537

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

The following dummy data frame has factor Q (with 2 levels) nesting factor P (with levels p1 and p2 nested under q1, and p3 and p4 nested under q2), but both crossing the random variate s, which has 8 levels. The dependent measure is dv. > # The data frame: > testnest dv s P Q 1 1 s1 p1 q1 2 2 s2 p1 q1 3 1 s3 p1 q1 4 2 s4 p1 q1 5 1 s5 p1 q1 6 3 s6 p1 q1 7 3 s7