similar to: Weird DTMF issue

Hi, I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original

This may not strictly be an asterisk question, but not sure where else to post ... I have an Asterisk test server setup with two firefly clients, one on the local lan and one on an external ip address. Both clients are setup the same way and voice calls work fine. The asterisk console reports a "Registered" message for the external client at about one minute intervals but the

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

G'Day, I have a bit of FireFly problem that hopefully someone has seen before. What happens is if I make to or receive a call from the FireFly network the call will connect successfully. However, around 10 seconds after I answer the call I am disconnected. The weird thing is same thing happens if I make a call. I've had a look at the * console and I can't see that my * PBX drops

I'm starting to get this to work! Well I got Voice Mail to work! All calls goes to voice mail without ringing the users phone (iaxComm). Here is my iax.conf and my extensions.conf Any help would be great!! Thanks -------------- next part -------------- A non-text attachment was scrubbed... Name: extensions.conf Type: application/octet-stream Size: 1039 bytes Desc: extensions.conf Url :

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

Hi, I'm trying to track down why I'm getting calls dropped on an infrequent basis between two asterisk servers which are at the same physical location and connected to each other with UTP ethernet. Here is the connection diagram Asterisk Server 1 ===UTPENET== Switch ====UTPENET==== Asterisk Server 2 I see sporadic RX and TX frame retries when I enable iax2 debugging on either box.

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from receiving inbound calls on my Teliax number. Outbound calls through the same server work fine, which rules out most networking

Hi, My asterisk server is behind firewall and i am trying to connect to FWD. i hv configured as mentioned in this link http://www.freeworlddialup.com/advanced/iax. i am able to register my server with FWD. But when i dial 393612, i always get 'No one is available to answer this time, try again later'. I hv portforwarded tcp 4569 and 5060 from my firewall to my asterisk server. Any idea

On the asterisk1 I got this: register => username:secret@ipaddress2 [eop] username=username secret=secret type=peer host=ipaddress1 auth=md5 on the second box I got this this host is ipaddress2 [incommingiax2] username=username type=user secret=secret host=dynamic context=from-internal-custom auth=md5 on first host 1 am getting: Jun 22 14:42:10 NOTICE[2398]: chan_iax2.c:7411

Hi, I have two asterisk servers and I want to make these servers call each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: ================== [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

Is there any documentation on IAX RSA authentication because I followed http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and it's not working... Asterisk 1 : -r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub -r-------- 1 root root 963 Aug 24 19:38 server1.key Asterisk 2 : -r-------- 1 root root 963 Aug 24 19:53 server2.key -r--r--r-- 1 root root 272 Aug 25 09:02

I have a PSTN termination provider "foo" which will accept standard U.S. calls in the form 1<10 digit ph#>. I have an outbound route named "foo", with dial pattern "5|.", with the only entry in trunk sequence being "IAX2/foo". I have an X-lite local extension, on which I can dial 51<10 digit ph#>, and asterisk will call out over foo and the