similar to: Advantage of IAX2 to SIP?

For efficiency & reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Thank you, Brent -------------- next part -------------- An HTML attachment was scrubbed...

What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider "callwithus" but they discontinue IAX2 I switched to "checkbox.cc" but they discontinued it as well. What is wrong with IAX2? -- Joseph

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

Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) Using the demo as an example, iax2 show channels Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter

Hi, I have some questions regarding IAX, IAX2 and encrypted authentication. How can I know if IAX or IAX2 is used between two * servers? There is any guide about how to configure encrypted authentication (not in clear text)between two * servers? I "hear" on this list a couple of days ago that port 5036 is the default one for IAX and something else (4XXX) for IAX2. Trying 'iax

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:

Hi All I'm trying to get two Asterisk servers to talk to each other using IAX(2). I've read the WiKi and the docs and tried the examples..... I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960 registering on the other). I've set them up as follows... The two servers are set up as friends and have consecutive IP address's. The setup is

Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when they call iax to sip, it works. When they make an echo test..no voice received on iax clients. When they make call from sip to iax ..no sound received on iax clients. I didnt see any clue on debug.

Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___________________________________________________________________________

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's

Is it possible to make Asterisk do inband DTMF over IAX?

Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S.

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any

I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to