similar to: IAX channel unreliable with multiple hops

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

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

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how?

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

Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing

Answer: Because it messes up the order in which people normally read text, especially the archives of mailing lists. Question: Why is Top-posting such a bad thing? Op 2013-11-09 om 09:22 schreef Jason Matthews: > On Sat, Nov 9, 2013 at 4:33 AM, Geert Stappers wrote: > > Op 2013-11-08 om 15:44 schreef H. Peter Anvin: > > > > > > syslinux.efi either crashed

Hello, The code here https://github.com/gsliepen/tinc/blob/master/src/graph.c#L150-L233 looks like the SSSP for indirect data forwarding is generated by looking for the minimal number of hops. Is there any possibilities or plan to generate the SSSP by using the weights of edges. And, is there any plan to add an option to allow indirect data forwards if the direct UDP channel is worse than the

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

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

Does the MWI feature work with IAX2? I have read where it should but cannot get the indicator to work on any of the IAX softphones that I have tried which have this feature. I even did an IAX debug and did not see where and indication was sent to the phone when it registered. IAX2 registration session: *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ

hi all, I'm trying to set up some iax2 trunks in Realtime architecture with the same backend. All work better (make call, receive etc etc) but when I do "iax2 show peers" some asterisk don't show anything and other show the iax2 peers but with status "unknow". Name/Username Host Mask Port Status ctm1/trixbox 10.0.0.131 (S)

Something I've just noticed that might persuade me to move to 1.4 ... in iax.conf, there is a new option: transfer=mediaonly Does this mean that it keeps itself in the loop as far as signalling/CDR is concerned, but lets the media stream go between the 2 endpoints? ie. Asterisk A <-> Asterisk B <-> Asterisk C Where B keeps track of CDRs, so right now I have

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

I have two asterisk servers connected using IAX. Server A has a TE410P running on a Xeon 2.4Ghz with 2GB RAM and 36G IDE HD on Debian 2.6.11-1-686 and Asterisk CVS-Nv1-0-7-06/01/05-01:27:25. Server B does not have any Digium board, but has ztdummy and zaptel loaded. It's runnin on a P4 1.6Ghz with 1GB RAM and 36G SCSI RAID 10 on Gentoo 2.6.11-gentoo-r9 and Asterisk 1.0.7. The

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let