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Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

Yeah, I'd like to hear you guys' opinion instead of CleverNickName's! -----Original Message----- From: Luki [mailto:lugosoft@gmail.com] Sent: Tuesday, February 15, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts?? You can always visit Slashdot for countless (useless, well, not always) comments: http://yro.slashdo...

I have many remote locations that dial into a central server to retrieve voicemail via IAX. Outbound calls are handled as SIP calls from a Snom to a local (to them) Asterisk server that dials the main server thru IAX. I have trained them to check their voicemail via the emailed WAV file, however some of them are, how shall I put it, idiots*, and insist that they *have* to have the MWI indicator

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone Due to security reasons I want to run asterisk as a non root. I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S

...reasonable fee doesn't actually include the thing working more than 50% of the time. -A. ANDREW - Thanks. I'd be happy to consider any US provider that can transfer / retain my POTS phone numbers. Any recommendations> RE Message: 7 Date: Sat, 7 May 2005 20:28:57 -0700 From: Luki <lugosoft@gmail.com> Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> My thoughts: 1) I would not run asterisk on a laptop, or on Windows (if you can get it to turn properly via emulation like...

Hi all, I'm having this weird issue that I can't explain. Maybe someone can explain what is happening. This is a Asterisk install that has been in production for 6+ months. It's version 1.2.10. Couple weeks ago one SIP peer started disappearing randomly. And I mean it simply disappears. One second "sip show peers" shows it, and then it's gone. A simple "sip

Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at