similar to: Recomendations for an ISDN-PBX to use with asterisk

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

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

hi All Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ? i need informations sample config for that, or can show how to route docs . thanks Dirgan --------------------------------- Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm analyzing freepbx extensions. When creating ivr with freepbx, it writes like this: exten => 1111,1,Answer exten => 1111,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID) exten => 1111,n(USERCID),Macro(user-callerid,) exten => 1111,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME}) exten =>

Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello flac-users, I'm trying to play an ogg flac file in WinAmp 2.80 but I get nothing. I'm using the native Vorbis plugin and the shipped FLAC 1.04 plugin. I guess it's a conflict regarding which plugin "owns" the ogg extension or format. I'm not sure. But I wanted to know how good the ogg framing was. But I couldn't test it because metaflac doesn't

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

As some e-mails ago, I'm trying to play beryl in my laptop. I successfully configured my xorg.conf (thanks to the comunity for the help that allowed me that). The problem: When I boot the laptop without external LCD/CRT, beryl is working fine, but when I boot it with the external, it recognice fine the external LCD and don't want to allow beryl run. What beryl says: === BEGIN === $

Right after I install the ruby and rails. I am able to use rails command. After I boot the computer. My window doesn''t recognice the rails or ruby command. I''m not sure if i need to run something first before I use the rails or ruby comment. Thank in advance. -- Posted via http://www.ruby-forum.com/.

Hi everyone I have a rather tricky question. We use skip 1.1.1 between our Suns to sign the IP trafic. A side effect of that is that the Suns can't see each other's broadcast. Now I want to set up two samba servers, with and without encrypted passwords. They must recognice each other, since one of the severs also is a domain master and they should both be annonunced in a different subnet.

This is a REAL problem on the way to world domination ;-) PROBLEM: since we moved a MS-Access MDB (16 Meg) to samba, it ist killing queries more than twice a day, loosing records, ... It seems the application has no chance to recognice locked records and inform the user. with smbstatus I get one locking entry for the first user accessing the database (like: DENY_NONE RDWR), the second is not

Hi all, I found a strange behavior in memdisk: I build an image of Windows 2000 (Fat32) with geometric of C=115 H=255 S=63. I testet it as a real harddisk and it works. I boot it to ram with memdisk initrd=w2000.img from my first harddrive, as discribed to boot with syslinux. It crashs with reboot, the time is to short to recognice the appearing letters before reboot. (Can I stop the screen?)

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with