similar to: Grandstream SYSLOG error codes

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the

Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in

Hello *, do queues allow me to set an announce like the A() option of the Dial() cmd? The announce that I've found is a message that is heard by the caller. I'd like to send a message to the member of the queue that picks up the call. Thanks in advance, -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l.

Hello *, I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens that I send a request and I receive a response to ANOTHER request that it got in the frame time between my request and my response. Did anyone else notice this behaviour? How can this be solved? I've been reading the source code, but I didn't find a solution. Thanks in advance, -- Dr. Andrea

Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success

Hello everyone, I'm Andrea, and I've started working with Asterisk a couple of weeks ago, so I'm still a newbie. :) I was reading about Asterisk Realtime, and I was wondering if I can mix Static realtime and "Real" realtime configuration. For instance: can I have a "Static Realtime" extensions.conf and use "Real Realtime" sippeers and sipusers? Moreover,

Hi folks, I'm really confused, so please help me, or at least give me some pointers to clarify this issue. Can I mix "Static" and "Real" realtime? Is there a way to easily switch from one to another, say, for sip.conf? Which are the major benefits of "Real" realtime? Please help me! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute

Hello guys, as you probably have already understood, I'm trying to make asterisk realtime work. Well, now it's working, but I'm not fully understanding the "metrics". In voip-info.org I found that they are a sort of "position" inside a context (var_metric) or the index of the context (cat_metric). Am I right? Where can I obtain more info about these

Hello everybody, I want to use the TIMEOUT() function, but in the CLI the "show functions" command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message "function LANGUAGE not registered". How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, --

Hello everybody, I'd like to hear some success stories about the use of Asterisk Realtime in medium-large contexts, like > 50 extensions. Don't you think that in those contexts the system could be overloaded from the excessive number of queries to the DB? So.. is anybody using ARA in those kind of deployments? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies

Hello everybody, I have a small problem: I've set "language=it" in iax.conf, but MeetMe conferences still play "en" files. I see from the CLI that the Playback app is called with "language=en" parameter. From the sources of app_meetme I see that it takes the language from the channel, so I think this is a IAX problem. Can anybody help me? Asterisk version

Hello everybody, I'm trying to understand how can I set the MoH class for parked calls. I set the "incoming" class for calls, and it works correctly. When I park the call, the music on hold is ok, but when I close the communication on the parking side, the parked call gets the default music on hold class. Can someone explain me what's going on? Thanks in advance, -- Dott.

Hello everybody, I'm aware that I can try to balance gain levels with PSTN cards using the ztmonitor tool, as described in http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html (Adjusting the rxgain/txgain Settings). Is there a similar tool for mISDN? If not, what is your approach to gain setting in mISDN? Thanks in advance, -- Dr. Andrea Spadaccini

X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. Not sure

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any

Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next

>I was successfully using the BT-100 phone with CVS 11/10. Now that I've >upgraded to 11/27, I can't place an outbound call. However the phone is >registered and works well with inbound calls. Any suggestions will be >appreciated. Thank you. Hi! I encounter similar problems. But in my case also incomming calls are not possible. But this might be because of my upgrade to

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I