similar to: Multiple incomming contexts

Dear list I have been at it for a few day trying to get my SuSE 8.2 server (FREESIDE) to produce a domain (SKOGFARET) for my Windows 2000 Server machine (BABYLON). I use OpenLDAP, and the setup seems to work in most ways, except the vital one [ie. I can mount shares, print, etc...] When I try to add W2K (BABYLON) to the domain, I get the following error: "The following error occured

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

Greetings. I have a Windows share mounted via Samba. I recently tried to rename some files; the destination filenames had some high-bit characters. The renaming had some completely unexpected consequences; the new filenames now contain the full pathname (including backslashes), plus other illegal characters (colons). The files can no longer be accessed or modified, either from the remote

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

Hi, I have two asterisk servers and I want to make these servers call each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: ================== [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal

Hi list I plan to implement a prepaid solution where the system needs to check for remaining credit periodically during a call. The reason for this is that this is a system where the credit pool can be used simultaneously by more people, and not only for calling. I have a problem figuring how to be able to run logic while a call is in progress. The L(x:y:z) option to Dial() is good, but not

Hi I have noticed new behaviour of asterisk 10.0 realtime. In 1.6 when I was using realtime: """ [somecontext] exten => someexten1...... exten => someexten2...... exten => someexten3...... exten => someexten4...... switch => Realtime/${CONTEXT}@extensions """ switch statement was executed after lines above (so there was a precedence of the

Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from

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 all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2

On Thursday 21 July 2005 17:42, asterisk-users-request@lists.digium.com wrote: > > > Thanks Adam. This helps some, but I'm still not sure how you mean ? > > > for me to acheive 1). I would have to perform a Dial-command no matter > > > what, so I guess I would have to make an interruption from the > > > manager API, but I don't manage to find a command

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

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

Has something changed in the recent modifications to Asterisk that would break dialing of the IAX peer? We're getting these authority failures everywhere. Everything is configured just the way it was half a year ago, this is the message we're getting on the most recent vers of asterisk. Wiki says nothing, nor does the ast-dev list.. -lost Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

I install the Asterisk into a RedHat9, exactly like manual says, and I'm having the attached error message when try to install libpri. Please, help on it. -------------- next part -------------- [root@LinuxLab zaptel]# cd .. [root@LinuxLab src]# cd libpri/ [root@LinuxLab libpri]# make clean; make install Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110309/fe9d7bc7/attachment.htm>