similar to: No sound

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing

Hi, What can it be when I can establish a connection between two Softphones but no voice is transfered ? thnx Hugo,

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255

I have installed Firefly, but I cannot figure out how to use it with Asterisk. I have seen the settings in Asterisk, but I do not see any settings in Firefly. I need a light .... bye Ronald

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ?

Hi Folks, After going to the paifull steps of installing AreskiCC and finally being able to access the webinterface, connecting to *, importing rates and setting up accounts I am not being able to make a CALL: No matter what number i try to dial I get the same response: The number you have dialed is currently unavailabel. Please enter thenumber you want to dial starting with 1 for local and

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

I have never noticed the message prior my upgrade of CVS head: chan_iax2.c:5589 update_registry: Restricting registration for peer '611' to 60 seconds (requested 1200) What does it mean, and how can I fix it? 611 is a firefly soft phone. bye Ronald Wiplinger

Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev & Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 & ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi, More questions in my ongoing quest to convert from RapidMiner to R. One thing has become VERY CLEAR: None of the issues I'm asking about here are addressed in RapidMiner. How it handles misisng values, scaling, etc. is hidden within the "black box". Using R is forcing me to take a much deeper look at my data and how my experiments are constructed. (That's a very

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Hi, For a few months i've been using WhiteBox Enterprise Linux. In fact, i've migrated all my companie's servers (about 20) to this RHEL distro like. Well, it's not that i'm not satisfied, but WBEL is a one man show and that causes me some preocupation. And besides, it seems to me that it can't stand up to the rythm RHEL evolves. That's why i'm writing to you,