similar to: Error with Connecting Two Asterisk BOX with IAX

Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hello all, We would like to "inform" the caller of the reason for a failed call. For example, when we get a "486 Busy Here", the system accepts it and in the CLI we see "Everyone is busy/congested at this time". Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael -------------- next part -------------- An HTML

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 Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set

hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko

Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes ; log date/time in GMT

hello, I' trying to use Asterisk 12 Alpha. Compilation and instalation without issues. When I try to start asterisk with: asterisk -cvvvvvvvvvvvvvvv i see this error on the console: 17:09:43.559 sip_endpoint.c !Module "mod-refer" registered asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. Any hints? Thank you

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

Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

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

Hello, I'm trying to use SIP_HEADER function on my dialplan but I receive this message (on the console): pbx.c:3367 ast_func_read: Function SIP_Header not registered Why? Thank's - Bakko

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

I posted earlier in the day about needed help with IAX trunking. I did some more reading and made some more changes. Here is what I have thus far: Iax.conf on one server [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming

Hi anyone know what is a the solution of this problems ? : [Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to [Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to 78.XX.XX.XX we have two Asterisk 1.6.1.4, this error are on the first server, used for the

Hello, maybe is a stupid question but I'd like know if I'm doing some mistake on IAX trunk configuration. I have two Asterisk (1.6.2.14) with this configuration: A (iax.conf) register => serverb:password at 192.168.142.246 [servera] type=friend host=dynamic trunk=yes secret=password context=phones deny=0.0.0.0/0.0.0.0 permit=192.168.142.246/255.255.255.255 qualify=yes B

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