similar to: - SOLVED - stream file not working but get data and exec background work

Hi I have a strange problem. I use the agi command stream file for my vertical services like *98 If the call comes from a sip phone with dtmfmode=inband in sip.conf then it works. But if I call the same script from an external line the stream file doesn't work properly The audio is played but the digits are not captured. I tested with get data and it works I tested with exec background

Hi Can someone recommend a PRI to SIP Box that work well with asterisk We are presently testing with a Patton Smartnode 2400 but we are unable to fax through it. We don't want to use digium card in a linux box for the PRI connection. Which Cisco box would work. Thanks Patrick

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

Hi We have static noise problem on our asterisk server. latest stable release. The card is a new TDM04B We have it installed on the following hardware Motherboard Intel SE7520BD2SCSI 2x POWER SUPPLY 730W INTEL I will not mention the other hardware because we have desactivated/changed all the other items The only 2 items that we have not changed is the mobo and the power supply. At first it

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC

This is what you show CBeyond. You have some vouchers there from CBeyond that will allow me to get paid from them not you. Chuck Coleman President CCI Technologies/CC Call Center/CSI Technologies Director of Managed Services for Gurus2go Cell 510-439-6501 Confidential Email: This email and any files transmitted with it are confidential and intended solely for the use of the individual or

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell

2010/10/24 Olivier <oza_4h07 at yahoo.fr> > > > 2010/10/14 Danny Nicholas <danny at debsinc.com> > >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier >> >> *Sent:* Thursday, October 14, 2010 3:34 PM >> *To:*

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PPPPPPPPPP] type=peer user=phone host=sip.broadvoice.com

Hello, I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38. My setup is as follows: SIP Provider --> Asterisk 13 --> Patton --> Physical Fax I need to get the fax directly in T38 to Patton. The provider sends me the fax in T38. If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax. If I send a T38 fax with Asterisk

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

Hi, I'm in the process of adding support for Markdown to a minimal CMS in Rails, [Railfrog][railfrog], which uses mime types to select appropriate processing. I have had a look through the archives but have not been able to see that a consensus has emerged as to what such a mime type for Markdown should look like. My reading of the RFCs suggests that it should be within the "text/*"

Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone> I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously,

I have been given a Patton SmartNode 4114 and asked to get it working as POPS gateway for our asterisk box. The 4114 has 4 FXO ports. It's got firmware 3.21 on it. I currently have a single POPS line plugged into port 0. I can not seem to get the two to talk together. I am running asterisk 1.2.21.1. I am seeing the following repeatedly in * Jan 24 16:23:40 NOTICE[17063]: chan_sip.c:11291

Hello all, I've began writing the parsing section of the spec, and I though I'd let you know about where I'm heading with all this. Basically, parsing is defined as three consecutive passes: parsing document elements, parsing block elements and parsing span elements. Each pass is going to contain a set of rules the parser should attempt to match while parsing the input. Rules

I am currently being defeated by grep. I am attempting to determine the value of the last letter of a character string. An example of my data set is shown below. Regarding the codes, I would like to identify the value of the last character and then take the appropriate action, e.g. If the value is L then label UL rating XXX It the value is F then label UL rating YYY ... I assume it will be

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton