similar to: Intel Vs AMD

I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric

I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is

Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building

Not being familiar with chip sets and other factors that "hardware experts" understand, I am perplexed with the multitude of available motherboards. Is there a Web site that can be of help or a "strategy"? My goal is to replace my motherboard with one that is Centos compatible and uses a Pentium 4 in the 2 MHz range. As example, when I go to Mwave, I am presented with

Hello, This about remapping in the HPA tftp server. A client does request for '/tftpboot/ncd4/foo' which should be 'ncd4/foo' I have created a remap file that looks now like this: r tftpboot/ncd4/modshmx/login.hmx ncd4/modshmx/login.hmx r tftpboot/ncd4/modshmx/setup.hmx ncd4/modshmx/setup.hmx r tftpboot/ncd4/modshmx/term.hmx ncd4/modshmx/term.hmx Because r

Anyone used one of these, or know anything about it? http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=AA74662 Breandan Dezendorf

Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I mentioned a ghost file. Here is the dump (less than 65M): http://dev.mccme.ru/~raskin/btrfs.dump ghost file is: 885dj1l4788pymp3bk2f3dz52ggcvw4v-empty/garbage-1/mwave.h.tmp-31838-1822528541 It has a name, but stat fails (so no inode, I guess). Also I can create another file with the same name in the directory. I cannot remove the file, or

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Dear All, I have installed 4 asterisks on the same Centos machine..>Each Asterisk has its own installation folder and use its own libraries...Everything looks great and all asterisks are doing their jobs correctly except one thing...I faced a voice quality issue...On a specific time, and after the number of calls begin increasing, the voice quality will begin degradation... Could it be a

Hi all, I'm using asterisk in real time mode...All extensions are defined in table sip_buddies...Everything looks fine and asterisk is reading extensions info from the sip_buddies table...The problem occurs as soon as any information on an extension is changed from sip_buddies table...Which mean, if I change the secret field in sip_buddies table then i should reload asterisk to read again the

Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Dear All, I have the below context defined in extensions.con: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Wait(2) exten => _X.,22,Record(/tmp/asterisk-recording:ulaw) exten => _X.,23,Wait(2) exten => _X.,24,Playback(/tmp/asterisk-recording) exten =>