similar to: Ser and Asterisk together

Hi, I can not send voicemails as an attachement. When setting the "attach=yes" option in voicemail.conf the mails get rejected from the mail server: ----- Transcript of session follows ----- 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed out with higgs.elka.pw.edu.pl. 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection timed out

hi all, I intend to write a snapshot module for Ext2/3 file-systems. For that i need to make sure that FS is in consistent state before taking snapshot. For ex. in case of UFS, the superblock has a fs_clean flag which is set to be FSSTABLE to tell that its in consistent state. (otherwise its set to FSACTIVE) I m not able to find any such information for Ext2/Ext3. (The

Hi, can any body suggest any tool which can show the utilisation for individual classes for HTB. preferable written in C/or shell script. Regards Jayesh ------------------------------------------------- Still single? Click here to find the perfect match. http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?141

Hi, More than 1000''s of classes and filters gets created/deleted in run time. Are we doing it correctly( as our requirement is such). Since few days we are facing lots of problems, like the server gets hanged, we get some junk messages in our logs. we have to reboot our server every now and then. What is the max limit of classes/filters can be created. we are using redhat 7.3.

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back

Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing

I don't know if this has been mentioned here or even thought of in the grand scheme of things and I'm just sending a one time email about it "just in case" I recently swapped my server around and I'm using dovecot now....LOVE it! Could use a bit more intuitive error messages, but otherwise, it's great! But there is one flaw that I've run into with my users.

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

Has anyone been able to change the way that asterisk performs transfers? Instead of using the # key, I would like to due something else, such as flash. # is just causing too many problems with transfers and menus when calling out.

Hi all, I would like to know if it is possible to use asterisk and ser together in a single computer system using ser as a sip proxy and forwarding any voice call request to asterisk for calling into the pstn gateway. (or any other alternative that is possible is also welcomed for suggestions). If it is possible can someone kindly show me the necessary configuration files or refer me to any page

Hi, I'm trying to use Asterisk as a SUA which would allow me to register on a SER running on the same machine and to send simple test messages through the CLI to another SUA located on another computer, registered on a SER running also on this other computer. I need the CLI because I have to work remotely. Is Asterisk the good choice? What kind of sip.conf and extensions.conf do I need?

I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive benefits, however, my initial playing around in SER's configuration indicates it's NOTHING like Asterisk at all, and almost 5x as difficult to understand and configure. But that's only after a few hours of playing with it. I'm interested in learning SER more, especially the integration with Asterisk. Is

Hi all, > >From Andy Powells Getting Started With Asterisk (V 0.1a) > > http://www.automated.it/guidetoasterisk.htm > > > > "" parking.conf file has this number set at 700. I've changed mine to > > 701 because I was having an issue with Asterisk - although it would > > 'see' > > (looking at the console) I had tried to transfer to

Is anyone else having problems with Sipuras not being able to re-register to Asterisk after applying the cvs update last night? Just curious if I need to roll back or take all of my Sipuras out back and beat them.

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan