similar to: SV: IAX - only one way traffic

Hello all! I?ve got a problem with the IAX setup. I?m previously only experienced with SIP, so that may be part of the problem. However, I?ve managed to register with the IAX server without any trouble (register line apparently works as it should), and I am also ale to make outbound calls. However, for inbound calls, all I get is this (from iax2 debug): Mar 29 17:44:18 NOTICE[11502]:

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

Hi, I created an iaxtel account, and was given a password containing an "@" character. The directory pages imply that they change the web login password only. How do I reset my IAX password so that it is usable in the iax.conf file? Thanks, Steve

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 want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

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

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 want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

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

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

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

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 :

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Hi everyone, THis is my second thread regarding the issue.(before I was having problems with accessing my email, which slow down my responses, sorry for that) My setup looks like this Firewall | | Asterisk---Asterisk (two asterisk servers with the same setup for high avail) | | phones Ports 5060, 10000-20000, 4569, 5036 are forwared to 192.168.1.251 which is virtual ip address on one of the

iax2 debug is giving following messages repeatedly. Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00001ms SCall: 00010 DCall: 00000 [xxx.xxx.157.230:4569] USERNAME : XXX9072835 REFRESH : 60 Tx-Frame Retry[002] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: PING Timestamp: 20001ms SCall: 00006 DCall: 00000

Hi! Sometimes IAX peers are not reachable and with "iax2 set debug on" I get something like this Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 00001 DCall: 01200 79.233.155.174:49153 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 01200 DCall: 00001 79.233.155.174:49153

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

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call