similar to: Problem w/ MoH

Displaying 20 results from an estimated 3000 matches similar to: "Problem w/ MoH"

2006 Apr 10
3
Vertical
Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other
2010 Oct 26
2
No media being sent in SIP call
Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone
2010 Apr 13
2
All incoming calls landing in [customers] context
Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls
2012 Apr 27
1
No UDPTL ports remaining
Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.
2012 Jan 26
2
Too many open files
Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket
2004 Dec 15
7
VoIP Termination
Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the
2009 Mar 13
3
Initial silence during call
Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike.
2011 Apr 25
3
PAP2T auto answer?
Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.
2010 Mar 29
3
Foip solution
Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from
2011 Dec 12
2
What version to upgrade to...?
Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd
2011 Sep 29
1
Features not working
Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl.
2004 Jun 23
5
Skype 4 Linux
Hi All, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation
2011 Jan 27
3
A1200P comments?
Hi all, Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card from OpenVox? I'll be using one to with 8-12 fxo interfaces.? The cards will be plugging into a cable-modem / phone adapter.? We weren't able to port the numbers, so we're going to use the existing PSTN connection and replace all of the office phones. With these short distances, will I need to worry
2023 Oct 10
1
Deleting voicemail by program
Here is something I wrote years ago. I expect you can adjust it for your needs # cat remove_blank_vmail #!/bin/bash # remove_blank_vmail takes arguments as voicemail boxes and removes messages with audio files shorter then MINSIZE (in bytes) #---------------------------------------------------------------------- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6
2023 Oct 09
3
Deleting voicemail by program
Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I
2012 Aug 01
2
Problem provisioning Cisco SPA303
Hello. I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that this device looks very similar to a PAP2T, so I used that as a template for my provisioning file. However, the result is less than stellar. Line 1 registers and works. However, lines 2 and 3 also register as line 1, effectively giving me a 1-line phone with 3 buttons. Also, the line name is the
2016 Apr 16
2
confbridge setup
Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be
2016 Mar 23
3
ODBC crashing asterisk
Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use
2009 Feb 09
2
SMS /w Asterisk
Hi all, I'm looking into being able to send/receive SMS messages with my asterisk box in the US. I've seen the SMS command as well as the Kannel program. I'd prefer to do it from Asterisk. I've tried something like: exten => 999,n,sms(15551234567,s,"This is a test") in my dialplan, but when this runs, it dials the phone number and then nothing. What am I
2007 Mar 28
3
Call dies when I press *
Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl