similar to: A problem with IAX2

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

B.H. Hello, all. We have noticed many calls on our PBX get "stuck" - the other end sends BYE, and our side sends ACK but the call remains active (no hangup event on AMI, the call is listed in 'core show channels') and it's impossible to hang up until asterisk is restarted. Asterisk's log shows lots of messages like this: chan_sip.c: Autodestruct on dialog .... with

B.H. Hello, all :-) We have a cluster of Asterisk (v. 11.9) servers that host IVR applications. The servers work behind SIP proxy (kamailio) for load balancing. All servers are in 2 processor configuration, 8-10 cores per CPU. When a particular server gets about 500 concurrent calls, the sound quality begins to degrade, the sound plays slowly and with clicks. As far as i understand, it's

B.H. Hello, all :-) We have some OPENVOX D410P PRI cards and we are successfully using them with Asterisk boxes which are based on stock ubuntu 12.04 DAHDI and Asterisk packages. The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it uses wct4xxp driver. Now, i'm trying to run this hardware with DAHDI 2.6.2 package which is available from asterisk.org site and looks

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. Our software is written

I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina 2770 1 Up Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request IAX2/from-CD-11006" several times but no joy. I also see the following in the CLI: [Nov 3

Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from receiving inbound calls on my Teliax number. Outbound calls through the same server work fine, which rules out most networking

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

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

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

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 hope you might give me some hints on how to find where my configuration is wrong, I am new to Asterisk and do not know, how to find the problem. Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the same maschine: Hylafax fax server. I want hylafax to use t38modem (a virtual T.38 modem) for sending faxes. t38modem schould connect to asterisk on the same host. If hylafax

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

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "door stopper"? -- Joseph

On Mon, 2 Mar 2015, Mordechay Kaganer wrote: > When a particular server gets about 500 concurrent calls, the sound > quality begins to degrade, the sound plays slowly and with clicks. As > far as i understand, it's because asterisk is unable to send the voice > stream in time i.e. the server is overloaded. > > What i don't understand is, at the time that the server

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

Hi all, Just hit this problem for the first time: WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from address 10.25x.xxx.160 When I ran "iax2 show peers" everything comes up as unreachable, no calls are passed between the servers (as would be expected) but there is no problem with DAHDI / SIP channels. At the same time there are no network issues (can ping all

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

Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965XXXX context=viatalk-in type=user