similar to: Dropping calls on IAX2

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all

I'm trying to figure out if this is a nat problem. I have a private network behind a freebsd nat box. The * server is on a static nat, with a private ip of 10.139.10.165. I'm connecting with sjphone as the client from 10.139.10.159. I am calling out using simpletelecom. When connecting directly to simpletelecom using sjphone everything works fine. When I go through * I get

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

Has something changed in the recent modifications to Asterisk that would break dialing of the IAX peer? We're getting these authority failures everywhere. Everything is configured just the way it was half a year ago, this is the message we're getting on the most recent vers of asterisk. Wiki says nothing, nor does the ast-dev list.. -lost Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

I spent the better part of the day trying to figure out why my SIP calls going through * were just going dead after 20 seconds. I was sure it was a nat issue but now I'm not so sure anymore. I have * on a public ip and clients behind a nat. I was using simpletelecom to terminate my calls. I could connect fine if I went direct from client -> simpletelecom. If I used * as a proxy the

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

Hi: We have got SIP clients connecting to our Asterisk fine with a DSL connection behind router (NAT), but when we bring the Sipura 2000 ATA to a Rogers Cable connection behind a Netgear router (NAT), the SIP clients aren't able to reach the Asterisk at all. We enabled the SIP debug in Asterisk, and it doesn't see any request coming from these SIP clients, and we also tried the to use a

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

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