similar to: tos problem

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Hello, Are there any T1/E1 PCMCIA cards available on the market supported by zaptel drivers and asterisk ? I need to make some demos at my clients with asterisk and it's a pain to move around with a midi-tower computer just for that. Thanks, Calin.

hello list, last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music on hold did not work anymore. my setup is ISDN (wct1xxp)->SIP (Audiocodes mp124) and reverse. the system refuses to activate music on hold resource... i returned to 1.0.5 and it works fine again... i'm i missing something? i can really make use of 1.0.6 bug-fixes and i'm sorry i can't use it :((

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

When experimenting with ToS, what would be the most appropriate combination to start with? I'm thinking "tos=0x14" should be good in most scenarios, since it combines "lowdelay" with "reliability". Any suggestions? -- Florin Andrei http://florin.myip.org/

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

I can't seem to change the default registration for IAX clients: Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'virbiage' to 60 seconds (requested 3600) Feb 6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'test1' to 60 seconds (requested 1200) Can this be controlled on a

Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin.

Hello everybody! I am having the following problem and since I am a beginner in playing with asterisk, i can't solve it: I am trying to integrate my existing H.323 network in real world telephony by ISDN cards. The problem is that i DON'T want to change all e164 numbers in my h.323 network and my ISDN provider doesn't accept those identities (CIDs). So, i have to spoof the outgoing

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip extension / sip softphone (linphone) 5. 1 x 800Mhz Asterisk + Linux server 6. Asterisk version is 1.6.2.13 7. 1 x IAX2 incoming trunk from phone provider for 1