similar to: Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si,

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover "up to 300,000 square feet". I am finding this hard to accept. I was also wondering about the "secure WDCT cordless technology" Could this be a form of DECT? Any one using these that can shed some lite?

Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar

Hello, I have a situation where a remote worker dials in to the asterisk server, enters the "secret code", then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] --

I am failing to get compactPDF to make any change to a pdf file that, a/c to the message from the CRAN upload site, can be very substantially compacted. Any ideas what may be wrong? I have also tried recreating the pdf file. I also tried R CMD build --resave-data --compact-vignettes DAAG The data files compact alright (but I get the 'significantly better compression' warning message

OK this is the RTFM question of the day but I need a sanity check. I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection. 2 Aastra 51i---------| |-NAT on dsl moden--(Internet)--Asterisk PAP2t----------------| The DSL modem/router which has QOS set for the src and dest to the * box the PAP2 has both lines registered @ ports 5060 and 5061 and work like a charm. one of the

Hi all, I am using * as a PBX for a Broadvoice VoIP account. It had been working well since about last November, although not perfectly (similar disconnection problems, although I am pretty sure it had to do with my PPPoE setup, but I think these issues were resolved). As of a few weeks ago, though, I started having serious problems. Basically, I can start up * and connect to Broadvoice and

Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get: