similar to: dialplan to try VOIP providers if they can't terminate call

Displaying 20 results from an estimated 20000 matches similar to: "dialplan to try VOIP providers if they can't terminate call"

2005 Sep 12
0
get dialstatus variable when returning No such context/extension
I have a list of VSPs that I use. Some are not able to terminate to different locations. It appears they are returning this error message: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension I would like to find out what the dialstatus is on this so I can try a different VSP that is able to terminate the call. Right now I have this
2004 Sep 23
1
video via IAX or SIP
HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes
2006 Jan 21
0
Dialstatus Oddity in 1.2
Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",
2006 Jan 16
0
asterisk 1.2.1 crashed
Hi guys, I'm using asterisk 1.2.1 since a week ago or so. today I found it crashed when making a call through teliax. This is how it looks: -- Called xxxxxxxxx@teliax/17075471770 -- Call accepted by 208.139.204.245 (format ulaw) -- Format for call is ulaw Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16
2003 Dec 18
1
Excessive VNAK's and jitter over IAX2
Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the
2005 May 12
2
UNREACHABLE messages
I get these on a consistant basis for most of the providers I have configured. Some less than others. I even get it from my phone at home to my * box at our data center. What I'm confused about is why it always shows the ping times at right around 2000 ms. That just can't be right. It's always right at 2000 ms. Never less or more by more than 100 or so. May 12 17:42:23
2007 May 14
1
IAX2 peer unreachable in one direction - NAT problem?
The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below
2007 May 15
0
IAX2 peer unreachable in one direction - NATproblem?
To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian
2005 Jan 27
1
Moh in meetme doesn't work if I transfer to meetme
Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:
2005 Sep 03
0
chan_iax2.c:7672 iax2_poke_noanswer
I have two units at customer locations in the Caribbean registering to a server in the US. Both units are connected to the Cable TV company's internet feed. If I run mtr to the units I see clean internet and low latency, but when I watch the CLI, I see constant problems. The audio quality is terrible, but I can't see why. Sep 3 16:42:46 NOTICE[20423]: chan_iax2.c:7014 socket_read:
2004 Jul 14
0
Originate to IAXComm problem once again
I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command
2004 May 23
1
IAX2 REACHABLE/UNREACHABLE
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,
2003 Dec 24
0
registration problem
Hi, Why do I get registration refused errors with Asterisk and voip providers? I did everything correctly and every provider I signed up with gives me that error: Dec 24 15:30:13 NOTICE[-1254995024]: File chan_iax.c, Line 3955 (socket_read): Registration of 'in-STn46BoD89' rejected: Registration Refused Dec 24 15:30:13 NOTICE[-1298793552]: File chan_iax2.c, Line 4389
2005 Mar 27
0
Voicemail / Dial command issue
Hi, I have a load of IAX extensions, which I'm trying to set up a standard macro to dial them, which gives unavailable or busy voicemail if there is no answer or the phone is in use respectively. The macro I have at the moment is: ; std-exten macro, ${ARG1} = Device to call, ${ARG2} = voicemail box [macro-std-exten] ; Call the user for 20 seconds exten => s,1,Dial(${ARG1},20,tr) exten
2003 Jul 24
0
IAXTel Connect Problem - Mini Frame
I'm new to the Asterisk software but have successfully set it up to make and receive calls using FXO cards, voicemail transfer etc. I can successfully call the Digium test IAX using the examples provided. I have signed up for an IAX tel account and got a number. The extensions have been set up as per the examples from IAX tel. However when I try to place a call this is what I get: --
2003 Sep 23
1
PROBLEMS WITH IAXATEL AND DIGIUM IAX
Hi.... I'm having a extrange problem.... I cant register with Iaxtel or call to digium... But i cant make or recive IAX calls... ( I made some one with irc users ) Any idea why? At my logs i have this from iaxtel: NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'xmarts' (from 192.168.0.11) NOTICE[196621]: File chan_iax2.c, Line 4389
2004 Dec 26
7
IAX Registration Refused
I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging
2006 Mar 29
2
IAX - only one way traffic
Hello all! I?ve got a problem with the IAX setup. I?m previously only experienced with SIP, so that may be part of the problem. However, I?ve managed to register with the IAX server without any trouble (register line apparently works as it should), and I am also ale to make outbound calls. However, for inbound calls, all I get is this (from iax2 debug): Mar 29 17:44:18 NOTICE[11502]:
2007 May 30
2
(no subject)
Need some help with IAX trunking. I've got six systems: AsteriskM (main) ___________________|____________________ | | | | | Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5 AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk boxes are using ztdummy for timing, they are all using IAX trunking. My calls come in
2003 Oct 24
1
IAX CALLS ONCE MORE
Hello, I updated CVS and nobody can call me any more with my IAX number 17007591228. I can only call other number but nobody can call me. This is what I get on debug when I call myself: -- Executing Dial("SIP/1011-7424", "IAX/bartosz:password@iaxtel.com/17007591228@iaxtel") in new stack -- Calling using options