similar to: Weird 7940 issue

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

Hi all, I'm having sudden MWI problems. Everything else on the phone works fine though. I have three Cisco 7940s. Asterisk server is behind a firewall running NAT. (192.168.1.202/24) Phone #1 - On the same subnet 192.168.1.250. Everything works great. Phone #2 - On a different subnet, 192.168.2.0/24. Everything works fine except the MWI. It never comes on. This is over an IPSEC VPN, but

Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas

Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

I frequently get this error message, it repeats itself hundred/thousands of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez

[answeringsvc] exten => 0,1,Wait,1 exten => 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r) exten => 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr) exten => 0,103,Goto(0,3) exten => 0,104,Goto(0,3) This should call 713-555-1212. If there are no ZAP lines available it should kick back around and play music on hold until a zap line is available, correct? I'd like the

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

Hiya, here is a patch to fix that: [root@mailmx2 channels]# cvs diff chan_vpb.c Index: chan_vpb.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v retrieving revision 1.9 diff -r1.9 chan_vpb.c 100,102c100,102 < static VPB_TONE Dialtone = {440, 440, 440, 0, 0, 0, 5000, 0 }; < static VPB_TONE Busytone = {440,

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

Hello. If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react. The messages from Asterisk in verbose mode: er will net. Asterisk messages in Terminalmode: parse_srv: SRV mapped to host sip-ha.web.de, port 5060 Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown"

Included 3 patches that updates documentation. This completes a 5 patches set. If you prefer me to resend all of then as one patch, attached, discussing or whatever, your're welcome. Best regards, vicente >From 7cec3ad78c8454408c8b6a1950d441e02d56d138 Mon Sep 17 00:00:00 2001 From: Vicente Jimenez Aguilar <googuy at gmail.com> Date: Fri, 23 Jan 2009 00:57:48 +0100 Subject: [PATCH]

Hey all. I've got a weird problem with the grandstream budgetone101 and asterisk that I'm having no luck finding any info on. I'm positive it's a grandstream problem but i'm hoping someone here can at least point me in the right direction. Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the

extensions.conf has ignorepat => 9 exten => _9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it asked if instead of keeping the same dialtone after pressing 9, if I could play a different dialtone. Can this be done? I'm running asterisk 1.0.0 in case that matters.

Folks -- I know this isn't directly an * issue, but I need to buy 14 7940s (preferably) (or 7960s if the price is also reasonable) --- no power cubes, immediately. If anyone has a good price, contact me offline at 512-427-1324 or lenny @ rocksteady.com Thanks, Lenny

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to