similar to: can't hear 'service messages' when iax is in the middle

I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : Input (client) -> Server (routing) -> Termination transfer=no transfer=mediaonly transfer=no all the machines are in the same 192.168.0.x net the routing Server in the middle has iaxusers realtime backend on mysql the call is originated with a sip phone registered on the Input client

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

I saw this thread which is very similar to my issue, though I cannot solve mine with iptables. http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html Using asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tables looks like this: root at dlaptop:/home/darryl# iptables -L Chain INPUT (policy ACCEPT) target prot opt source

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

I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. When I start the IAX Client the CLI show me the message: -- Registered '2004' (AUTHENTICATED) at 192.168.199.69:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home at my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

Hi to everybody, My IAX is not working, When I type reload IAX it returns me: AsteriskSlave*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config: Ignoring bindport on reload [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config: Ignoring bindaddr on

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

Hi, not sure if it's possible to do something like this. My main * box here at home is sat on the local lan, 192.168.1.0 , also on this lan I have another * box which is running as part of a wireless Meshbox (http://www.locustworld.com) . Now then, this meshbox has a wireless connection to another meshbox at a friends house. These meshboxes use weird IP addresses in the 1.x.y.z range

Hello, I'm using asterisk for a quite long period, i integrated a lot of stuff to make it behave like any carrier class system, so users can: manage Call forward on busy manage Call forward on no answer manage unconditionnal call forward call back missed calls and a lot of such services that can be both configured by menus and dtmf or by web interfaces (PHP+SQL), in fact i recreated

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

I have, in iax.conf the register statement: register => username:password@iaxtel.com This causes registration attempts to iaxtel.com for both IAX and IAX2. Every once in a while there is a packet for port 4569 keeping the IAX2 registration alive. This is fine. But, I have a barrage of registration attempts to iaxtel on port 5036 for IAX. Every UDP packet is answered with an ICMP packet

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other