similar to: Gatway with IAX ?

I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and

I find IAX connection with FWD very unreliable so I think I'll have to roll out my own "SIP Express Router" as I want to communicate with few SIP clients. So I hope this the right solution. I'm new to "SER" and to my understanding "SER is like a "road-map" it tells the SIP Clients where they are so they can communicate directly with each other without

I'm just curious why Sipura isn't using free IAX protocol with their devices instead of SIP? With IAX NAT traversal would have been easier, so why are they using SIP. Is there any politics in it? -- #Joseph

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel? There is a new application from Newman Telecom for fax detection. http://www.sineapps.com/news.php?rssid=575 Current Asterisk Fax detection doesn't work for me as I don't have Digium cards; I'm using Siupra -- #Joseph

Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.

I'm testing Teliax tall free number line and I'm experiencing long delay about 1sec. during conversation. When I call myself over FWD the response is normal no delay or cut messages. When I call my number over FWD the is a long delay, welcome message usually cuts off few first words and during conversation my voice arrives after about 1sec. delay. Since, the 800-number is only accessible

When I try to call out to FWD over IAX2 I get: Call rejected by 65.39.205.121: Unable to negotiate codec I'm using asterisk-1.0.5 (the same settings works fine with *0.9) I've standard settings in iax.conf [general] bindport=4569 register => xxxxx:xxxxxx@iax2.fwdnet.net [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw -- #Joseph

On Sat, 2005-06-18 at 13:17 -0700, Matthew Asham wrote: > For what it's worth I use a residential Shaw connection and have no > problems with IAX registration nor SIP to our Asterisk PBX elsewhere on > the 'net. When I had a dynapic IP connection with shaw, there was not problem either. I just changed to static and IAX can not connect. How can I test it? iax2 show registry Host

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

When loading iax.conf I get warning: WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now In iax.conf I have: [general] port=5036 -- #Joseph

What dtmfmode should I set for IAX protocol? When I dial FWD over IAX it doesn't recognize the numbers I press. -- #Joseph

Can anybody share information how to install NVBackgroundDetect? I have the app_nv_backgrounddetect.c but I'm missing: app_nv_backgrounddetect.o and app_nv_backgrounddetect.so and have no idea how to generate them. Wiki point to contact Newman Telecom but all I received was the app_nv_backgrounddetect.c and no instruction how how to install it. The installation instruction from wiki

Who else makes Analog Telephone Adapters with IAX protocol besides Digium? I've seen Farfon is advertising their unit but I'm not sure if they shipped it or not. Why is so hard for others to support this protocol in their adapters? SIP is totally unsuitable for firewall traversal for security reason. I would like to get min. two port unit. Is it possible to use IAX protocol without

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Is anybody using Asterisk with a dual WAN router (Xincom XC-DPG602, Hawking H2WR54G, Fortinet FortiGate-60, SonicWall etc) ? -- #Joseph

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? -------------- next part

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want