search for: ctjones

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Has anyone experienced any date/time problems on Asterisk? I have noticed it on earlier builds and I believe on this build: CVS-02/16/04-23:57:06 Specifically, I have seen the following problems: 1. Change system time using the "date" command while Asterisk is running. Asterisk does not synchronize with new system time. 2. Daylight savings time goes on/off at the designated

I am wanting to perform toll bypass using multiple gateways for outgoing calls. For example, if I call from location A to location B and I have a gateway in location B I obviously want to use location B's gateway to make it a local call. I understand how to get the local prefixes from NANPA but my question is: If there are additional numbers that are local (non-toll) calls from a particular

I am a fairly new user of Asterisk and I am generally impressed with its features. I have some questions about the SIP channel support: 1. I have noticed that even when there are no active calls, there is a list of active SIP channels. This appears to be a bug. Has anyone seen this? 2. If there are stuck SIP channels, how can one clear them without re-starting Asterisk? 3. One time a

I have setup Asterisk to work with a SIP gateway, some SIP phones and the Digium FXS/FXO development card combo on another * box with pretty good results so far. Here are a couple of questions that I have that wasn't obvious from the documentation: Voicemail vs Voicemail2 - What is the major difference between these and does Voicemail2 use different parameters or config file options?

I was looking at some fixes in the replies to the chan_sip.c problems and I am wondering if I am seeing the same thing in the earlier file version. I just checked to see that my chan_sip.c is version 1.179 when I did my checkout so I never had the later versions. The problem that I am seeing is that DTMF is not going from 1 SIP device to another and sometimes voice is not going from 1 SIP device

What is the difference between the version 1 and 2 voicemail and IAX? I see different examples using both so I want to know why would one use a particular version. Thanks!

I am relatively new at Asterisk but have a 2-machine system running with the developer kit (FXS/FXO cards), some Cisco SIP 7960's and a Audiocodes MP-104 SIP gateway. I would like to fix a couple of the voicemail boxes so someone can press some numbers such as "911" and get sent to a "priority" voicemail box. This different voicemail box would weed out the normal calls

I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would

Is there an option to configure Voicemail2 to NOT store the voicemail messages on the disk once they have been emailed to one's mail server. It can be a pain for some to receive voicemail via email and then go to Asterisk just to clean out the voicemail you just heard.

I have finally crashed Asterisk for the first time and I'm wondering if anyone has seen this. This is a configuration with SIP endpoints and an IAX2 channel to another Asterisk PBX. The main PBX dropped a core file after a SEGV (signal 11 ) with the following trace: #0 0x42079133 in strchr () from /lib/tls/libc.so.6 #1 0x41bb0f9c in _fini () from /usr/lib/asterisk/modules/chan_sip.so #2

I have had several cases where the message waiting indicator was stuck in the on state with Cisco 7960 SIP phones. Here are the two cases: 1. Single extension that mapped to a single voice mailbox. Restarting Asterisk or getting a new voicemail then clearing it fixed the problem. 2. Three SIP extensions that mapped to a single voice mailbox. Getting a new voicemail and then clearing it

I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have

I had a disturbing problem with my G.729 licenses the other day and I'm hoping that someone can provide me with some insight since Google didn't. I have 2 Asterisk boxes, 1 acting as the main server with 10 G.729 licenses and 1 that is acting as a media gateway with 2 G.729 licenses. The gateway build is: CVS-02/16/04-23:57:06. I have a TDM400 card with 1 FXS board (unused) and 1

Has anyone found any good online resources for performing transmission tests for POTS lines? There is plenty of info on this list about adjusting gains on X100 cards, etc. but I am looking for test procedures using test sets. I'm talking about tests for echo loss, distortion, etc. Thanks in advance for you help!

I am trying to music on hold but I am having all sorts of problems with it. I am running RH9 and the latest version of Asterisk as of yesterday. Here is what I did to test it: 1. I manually deleted the mpg123 softlink to mpg321. 2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed the the archive and loaded ztdummy.o module. 3. I threw a couple extra files in the mohmp3 directory

I have a friend with a PRI coming into a modem bank that is receiving 56K modem calls and some ISDN data calls. He wants to dump his analog office phone lines and use some of the capacity on the PRI. I have been digging through the mail archives and Wiki site on this subject but the information I found doesn't give me a high enough confidence to go buy a few T1 cards and try it out.

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most