similar to: T38 problem

This is one long letter about T.38 and Asterisk. I hope it will help me, and lots of other Asterisk users to understand some T.38 problems with Asterisk. This is my situation: I have Panasonic DX600 FAX machine. It's connected to Asterisk 1.2.13 thru ATA adapter (I have used both, Cisco 186 and Grandstream HandyTone 386). Asterisk is connected with my SIP provider. That link that my provider

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Hi people, pls does anybody know what "(T)" and "(D)" letter means? server3*CLI> iax2 show peers Name/Username Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? ---------------Logfile-------------------------------------------- Nov 14

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T ----------------- Certified Networking Professionals make better Connections! http://omarept.blogspot.com/

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded

Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just downgrade to 7.5 and give it a go? I think SIP uses UDP 5060 correct? The phone is behind a

Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 -------------- next part

I have been forced to introduce a Cisco 7971G-GE into my network, because "it has a pretty screen." I have wasted nearly three days fighting with the thing based upon the information on voip-info.org and a few other forums. Asterisk is reporting a 401 Unauthorized. Which typically means bad username/password combination. Unfortunately, all of the usernames and passwords I see in

Hi list, i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this? CLI: -- Accepting UNAUTHENTICATED call from 192.168.0.2: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (g729|ulaw|alaw), > priority = mine -------------- next part -------------- An HTML

Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but

Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards,

Hi. I'm having problems with chan_mISDN configuration. Line "echotraining=yes" causes warning, when Asterisk is parsing misdn.conf and I'm confused why the PBX doesn't accept the setting. No matter which section I try to offer it, it is always invalid or out of range. The setting itself is supposed to be valid, it is in the sample configuration file of chan_mISDN 0.3.1.

Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

When I start * I get 6 mpg123 processes start as well. Is this normal? Often after a couple of days these mpg123 processes start to consume cpu and I have to kill them off. I do not have a sound card in the server and I have noload => chan_oss.so Simon