similar to: register_verify defined in 2 files?

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

I did some searching in the archive, but found only one message with this same question and no answer. Hopefully it's a simple config problem. When the Caller-ID is delivered, it is surrounded by double-quotes, like this: "ATA-57 1" On long caller-id strings, the last character is cut off to make room for the leading double-quote: "BudgeTone 1234 instead of BudgeTone

Apparently asterisk cannot reboot gracefully (unattended) when using realtime.... MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug. Since asterisk starts before mysql it will never start any workaround for this? g Regards Greg Cirino

It's been hours since I've seen a post from this list Must be broken again. Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to

Hi all, I don't know why but my * is not accepting Register Messages. Have you seen this kind of problem before?? I need help! Thank in advance. #################################################### *CLI> Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869 register_verify: Peer '111' is trying to register, but not configured as host=dynamic Nov 19 15:42:11 NOTICE[12893]:

Has anybody been able to get in touch with anybody at digium today? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Is there some reason the sipbuddies table structure was designed with sip config values as column names? Doesn't look very flexible It really should take the form of ast_config so when a new sip feature is implemented, you don't have to re-write the entire data structure too. Regards Greg Cirino

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to

Hello everybody! [Introduction] This is a quite long message, but I think the problem is interesting. [The problem] Does anyone know how can I tell Asterisk that a certain user has a certain telephone number (or address)? For example, I have some registered users, but nor the client (X-lite) nor the server (Asterisk) specifies what telephone number has the user. I don't want to

My asterisk process produced the following errors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions (SIP) configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and