similar to: call to a particular 800 number nevershowsanswered on Zap channel

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

All, Having problems terminating to a Nuvox PRI, the tech at Nuvox is saying Asterisk is transmitting in CCITT (aka ITU?) when they're expecting (and will only accept) ANSI. The question is, is there a simple way to change this or am I stuck with rewriting code? I googled and checked the mailing list and found nothing, I could be barking up the wrong tree I guess. PRI is not my forte.

Recursive indexing can cause R-1.8.0 (and R-1.7.1) to segment fault First of all, many thanks to the R team!! R is really a software for everyday work. Yes, I've found a fault, but I hope it's not just faultfinding ;-) In the NEWS file of R-1.8.0, first printed in R NEWS 1.7.1, there was given a promise: [ 1 ] o Recursive indexing of lists is allowed, so x[[c(4,2)]] is

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

I have an application written in c# which uses two instances of the theora encoder in a single filtergraph. Each one is followed by an ogg multiplexer for output to separate files. When I run the filtergraph the smaller video's buffer is intermittently written to the start of the larger video's buffer. I am using the latest filters from opencodecs_0.84.17338.exe and the smaller video is

I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

Failed to deliver to ''menyhert@one.net'' SMTP module(domain one.net) reports: host smtp-in1.nuvox.net says: 550 5.0.0 <menyhert@one.net>... User Unknown -------------- next part -------------- Skipped content of type message/delivery-status-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/rfc822-headers Size: 476

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

I am using the directshow Theora encoder to compress a live video stream. A dump of the encoder settings is given below. The starred settings have been explicitly configured and the others are defaulted or derived values. These settings produce an even, dark (almost black) recorded result as does any recording with Keyframe Freq <= 32. Keyframe Freq >32 works perfectly. The incoming

My apologies if this is considered somewhat off-topic but I presume that this information may be may also be of use to others using these filters. Is there someone with a better understanding of COM than I have who can tell me where I'm going wrong with the following C# interface definition? I can set and recover a quality setting using this interface but not the isUsingQualityMode flag. Not

Hi, I've tired to forward a Fax from Asterisk to Hylafax. It works so far until I tried with a Fax machine. I just got error shown in the log below. I'm not sure why. I've tested it with other 6 machines and they all work fine. Do you have any idea why? Pim Hylafax Session log: Apr 12 11:16:48.82: [ 5933]: SESSION BEGIN 000000078 492212601860 Apr 12 11:16:48.82: [ 5933]:

Hello list, I've configured dovecot-2.3.0 on CentOS 6 with the following options. CFLAGS="-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fno-strict-aliasing"? ? \ LDFLAGS="-Wl,-z,now -Wl,-z,relro"??? \ ./configure???????????????????????????????????? \ ??? --docdir=/usr/share/doc/dovecot-2.3.0?? \

Hi, When is start one vpn i get the following result: tinc10703003005 Link encap:Ethernet HWaddr C2:F7:7B:75:47:1A inet addr:192.168.3.20 Bcast:192.168.3.255 Mask:255.255.255.0 inet6 addr: fe80::c0f7:7bff:fe75:471a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most