similar to: IAX Registry problems

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the

I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser

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

Hello, I have some problems in placing calls through IAX... It does not work :) in the asterisk console I can't see nothing about dialplan enter or so, IAX debbugging seems to be unuseful... this is my configuration: [612] type=friend secret=123456 notransfer=yes disallow=all allow=gsm allow=ulaw allow=alaw context=from-internal host=dynamic requirecalltoken=no I enabled IAX debugging, but

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 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] --

Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register => coiax:pass1 at 69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:4569

Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the "registration" to send iax/gsm calls to System A

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

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 all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is

Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it can't find channel 35 but 35 belongs to my wctdm.

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

We have a problem with our iaxclients. Our asterisk runs on a public host with debian and many of our IAX2 clients are natted. The iax.conf looks like: [23456] accountcode=123 type=friend context=user auth=md5 secret=xxxx username=23456 callerid=Testuser 1 <23456> notransfer=yes host=dynamic The cli command IAX2 show peers shows all clients as unmonitored CLI> IAX2

Hi, I have a nufone connection (IAX2), works fine. In my iax.conf I do not specify a time interval that * needs to renew registrations with nufone server. However I can see following registration messages on my cli every 90 seconds (approximately) --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569 --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569

Hi. Probably been asked before, but my IAX provider assures me its not their problem.... I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them. IAX debug output is: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 00003

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

Another might be to make sure iptables isn't blocking the connection. You can run iptables -L -n -v To see if its set to block any ports. On June 5, 2017 9:06:55 AM EDT, thelma at sys-concept.com wrote: >I'm getting: >netstat -a |grep 4569 >udp 0 0 0.0.0.0:4569 0.0.0.0:* > >Should I be getting localhost IP? > >Thelma > >On 06/05/2017

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend