similar to: IAX trunks and audio codecs

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

I just found an old bug report at bugs.digium.com with exactly the same problem. It's really too bad this bug wasn't addressed: http://bugs.digium.com/view.php?id=7315 --- On Fri, 1/9/09, Vieri <rentorbuy at yahoo.com> wrote: > I'm trying to figure out how to reload iax2 (without > breaking existing calls) so it can listen on a new IP > address (like "ip addr

    I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them.  Calls come and go but there is no CallerID from the remote server either way.  One of the servers is running Asterisk 16 and the other is an older 1.8 install (I know, I am trying to get permission to update).  The trunk between servers is very simple.  Something like: Server 1 (Mexico) [panama]

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax

Hi, I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. I've created the requirecalltoken field in my (Postgres via ODBC) database, type text, and have variously tried it with 'yes', 'no' and 'auto' in the field. But the setting never seems to be used and

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

Hi, I'm trying to find a way to implement the following: I have 1 media source (IceS or MPD) and 1 Icecast stream (say, LAN radio). Once in a while I'd like this stream to be interrupted by short announcements (PA system). Input for these announcements can be from another source (IceS, MPD, Asterisk call). Anyway, to make things simple: I'd have dir1 with ogg music files for

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

My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef

I had two asterisk servers connected with each other, both were 1.4.22 but I've upgraded one to 1.4.37 and now I get a message when I try call from asterisk-1.4.22 to asterisk-1.4.37 ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.140.1 in the calltokenoptional list or setting user guest

I'm trying to figure out how to reload iax2 (without breaking existing calls) so it can listen on a new IP address (like "ip addr add local ..."). This alias IP is added/removed by a custom process (script) for clustering purposes. The iax.conf file contains "bindaddr=0.0.0.0". I tried a "iax2 reload" (executed without errors or warnings) but I'm still not

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

Hi i have a small problems on two Asterisk Server 1.6.4 : The first sent the call to the second, and in the second, i have a error : [Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address IP_FIRST_ASTERISK in the calltokenignore list or setting user 04TELNUMBER requirecalltoken=no on the second

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken

Hi, I'm wondering if someone could tell me if DNS names in Asterisk .conf files (eg. iax.conf, sip.conf) are supposed to be resolved only once at * startup or periodically, according to DNS timeouts. I'm asking because I defined host=voipb1.mydomain.local in a iax.conf trunk and it was working fine until the DNS A record of that host changed IP address. Asterisk still tried to send calls

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

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