search for: emmegi

I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

...one will automagically rings playing the announce "from-queue" and bridge it with the call that's waiting. So, I'm correct? Anyone experienced that or could give me a better way to handle that? Thanks a lot, matteo. -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

...but then it seems suppressed. here's my modem.conf [interfaces] context=telecom dialtype=tone mode=immediate features=dtmf,noquelch driver=i4l group=1 msn=xxxxxxxxx device => /dev/ttyI0 device => /dev/ttyI1 ingoing calls are ok. any hints? Matteo Brancaleoni mbrancaleoni at espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

...terisk... don't know if could be dangerous). Or the log could be rotated without any HUP ? Something like /var/log/asterisk/debug { missingok postrotate compress /bin/killall -HUP asterisk 2> /dev/null || true endscript } Matteo Brancaleoni mbrancaleoni at espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

...sers But shows channels says: Zap/10-1 (local s 1 ) Down Bridged Call Zap/2-1 Zap/2-1 (interni 13 1 ) Up Dial Zap/13/|20|m 2 active channel(s) And the call is present between the two. Any idea? Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

...e rem. console it does nothing and the console simply stop responds. I can type commands but don't get anything back. I must quit the console and get back to have it fully working (but the channels are still up, even if they are down (sip, zap and so) Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

Hello, For those interested in overcoming the problem with some NATs and Firewalls in regards to tftp. I found a nice little tftp server here: http://freshmeat.net/projects/jtftp/?topic_id=87 I tried it and it works great. Regards, Andres. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

...e , but was wondering if digium staff could start one from the official cvs rep . Or let someone start it, if they enable rsync mirroring of the cvs itself (also useful for backups ;-> ). Any other idea, question, comments? Matteo -- Matteo Brancaleoni <mbrancaleoni@espia.it> Espia - Emmegi Srl

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

...ah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch could do ;) if for the list is ok, I'll submit to the bug tracker, under a feature-request. Matteo -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl -------------- next part -------------- --- asterisk/apps/app_dial.c 2003-10-08 00:05:43.000000000 +0200 +++ dial-asterisk/apps/app_dial.c 2003-10-08 00:04:20.000000000 +0200 @@ -337,6 +337,7 @@ struct localuser *u; char info[256], *peers, *timeout, *tech, *number, *rest, *cur; char priv...

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James

...DM40B and noticed that when the kernel module is loaded (so the card is powered on), the chip onto the ringer module (the one below the S/N label) is pretty hot (I can't touch it for more than 2 seconds). Is that normal ? Or too much hot could be dangerous ? Matteo Matteo Brancaleoni Espia - Emmegi Sys Admin http://www.espia.it

Hellooo... Is it possible to fill a variable of PHP-based-AGI-script from dialed extension ? This is what I need to achieve: If someone dial an extension, say 777, I want the dialed extension (777) be filled into PHP variable. I need the dialed extension become a condition of PHP script. Help please... Thanks romsun _________________________________________________________ This mail sent

Hello- Is it possible to set the CDR record field called "accountcode" from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com

Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there.

Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb

Could someone point me in the right direction for setting up the mysql cdr function Thanks robb