similar to: Play Music on hold until a ZAP channel frees up.

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'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

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

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

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

Hi all, On my 7940 phone when I dial out I press 9, then the number. After I press the second number (IE: 9,1) the dialtone stops playing just like it should. This is normal and similar to a regular phone. On two of my 7940s the phones continue the dialtone. No matter how many numbers you dial the dialtone does not stop until you press dial. Also, on these two phones a little X appears next

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

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>

Remember in the good ol days when answering machines were smart enough to know when there was a message on the machine, and it would pick up after 2 rings rather than 4? (that is, if you knew how to turn it on - that required to know how to set the time on your VCR to avoid the flashing 12:00:00) Hahaha. Jokes aside. I have come up with a way to do this but it's a kludge: 1 - Read in the

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

i tried using music on hold option in the dial command exten => 7777,1,Dial(zap/1/7777,20,m) when someone calls me and i picked up the phone, the call will be suddenly dropped. however, if i use a sip client instead of zap (also changing the dial statement to sip), i can answer the incoming call without a problem. is this a known bug? (asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX)

Is there a way to play music on hold for a specified amount of time while listening for DTMF? I suppose I'm looking for a hybrid of Background() and WaitMusicOnHold(). I don't really want to use Background() because the music would start over each time.

Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I

I have an application where we receive calls on an inbound PRI. After hours, our Asterisk box dials our answering service on an outbound PRI and then bridges the caller to the answering service. The flow looks like this: (CALLER)INBOUND_PRI --> CONTEXT --> GOSUB(Incoming) --> GOSUB(bridge-to-anssrv) --> DIAL(answering_service) --> OUTBOUND_PRI(service) This has been working

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

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]

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a