similar to: help

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

At approximately 3:15pm I shut down the office MySQL server to change out some hardware. Shortly after I received a call from one of two customers whose asterisk servers output CDR data to that server. They could not place or receive calls. Shortly after that I received a call from the other customer. I'm below providing output from the message log (At debug level). I don't see much

Hi there: I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4 box. When I tend to install the addon for mysql CDR billing, It always return me the following errors: SIP# gmake clean rm -f *.so *.o .depend gmake -C format_mp3 clean gmake[1]: Entering directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' rm -f *.o *.so *~ gmake[1]: Leaving directory

hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get

I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

Hi to all!! Is there another method to download asterisk addons??? Thanks.... F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040528/8f1f7764/attachment.htm

I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?

Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that "soft hangup" should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command "iax2 show channels") IP-AM-PBX*CLI> iax2 show channels Channel

Hi all, I have a strange problem, regarding zap channels and cdr. I am using asterisk bristuffed version Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium, Inc. and others. with two billion ISDN cards. I also installed asterisk addons, last stable version via cvs internal calls, or calls starting from internal sip or iax phone are recorded in the cdr all without any

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hi, I need help with modifying cdr_addon_mysql.c I want to have more fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c and replace userfield with ex team (sed -e 's/userfield/team/g' ). When I try to recomplie menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'.

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

[ Sorry, I incorrectly copied some Reference headers into this post and tacked it onto the wrong thread. -Steve ] So far, I have been able to receive incoming iaxtel calls via my assigned 1-700-xxx-xxxx number, but only when using md5 authentication in iax.conf: [iaxtel] type=user ; Incoming calls only context=incoming auth=md5 secret=<mysecret> ; Required for

I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58 I get the following errors: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function

I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names