similar to: Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ?

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

Hi, Has someone met success setting a Patton M-ATA to work in T.38 ? In my experiences here, it seems this ATA don't switch to T.38 whenever a fax signal is heard on its FXS port. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090205/79d0a70f/attachment.htm

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

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 MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting "No signal. Line is

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hi, I can not get this working: Asterisk on public IP. Two SIP phones behind NAT - in the same LAN. I works perfectly (two way sound) when each peer (friend) can not reinvite - audio stream goes through Asterisk. The problem pops up when I define canreinvite=yes on each peer definision so I suppose to stream audio directly between phones (in the same local LAN). Right after called party

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 All, Currently I have voice calls from a certain SIP peer coming into an asterisk server where the specific [SIP] channel is set to 'canreinvite=no'. I would like to enable reinvites for certain calls, matched on DID. So I'm wondering if there is a mechanism in the dial plan to turn on/off reinvite capability or will every call on this channel be forced to use the SIP peer

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other. jtodd has had this problem in the past with the 186's. Just wondering if anyone has a reason why "Cisco sometimes poop on reinvite" is the Cisco code broke? if so we can push on Cisco to fix it. the U is a MAJOR Cisco shop so we have some puhs

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been

I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just "yeah, it should work.....". If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong,

Hi, Sorry for this "newb" question (but maybe a newb question once in a while is ok): What's the current state about Asterisk handling DTMF features via SIP INFO (dtmfmode=info) even when the media path has been reinvited (canreinvite=yes) to go directly from one phone to another? Somewhat related to this suspended issue: https://issues.asterisk.org/view.php?id=14126 How widely

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

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, 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

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

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax