similar to: Faxing with Asterisk 1.8.4 & T.38

Displaying 20 results from an estimated 5000 matches similar to: "Faxing with Asterisk 1.8.4 & T.38"

2005 Jun 24
4
Tellabs Echo Canceller
I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive
2005 Aug 26
2
Help Solving Asterisk Lockups
I am currently testing a new Asterisk installation. The server has a T100P connected to a PRI, and about 50 Polycom IP600 phones connected via the local network. Every couple hours, Asterisk randomly stops responding to all calls, both incoming on the PRI and calls from the SIP phones. I'm not sure how or where to start debugging it. When Asterisk stops working I can still connect to
2007 Jul 18
1
Any way to determine remote Asterisk version
A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant
2006 Nov 29
2
Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other
2005 Oct 05
3
IPComms Setup
Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect
2009 Jan 17
3
Asterisk 1.6 T38 to G711 transcoding is this possible?
The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and
2004 Jul 28
3
faxing
What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.
2004 Dec 07
1
IAX DIDs, Illinois
I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up
2006 Feb 02
0
Anyone know a good ITSP in Canada that suppo rts *?
There are a number of them, try Comwave, Voxip or Wiztel. Depends on what you need we may also provide it... email me privately if you're interested. Some provide IAX, some only SIP, H323, & MGCP... -----Original Message----- From: hugolivude [mailto:hugolivude@gmail.com] Sent: Thursday, February 02, 2006 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:
2011 Jun 21
1
: Re: ITSP failover for PRI
Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I
2007 Dec 10
1
T.38 fax solution, opinions?
Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by
2016 Nov 15
2
iaxmodem errors.
2006 May 25
2
Volume configuration on Polycom Soundpoint 501phone
Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >
2010 Jan 04
1
T.38 ITSP?
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send
2008 Aug 15
2
DID's needed for Reston Virginia - + hosted asterisk
I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be
2005 Feb 25
4
T.38 fax summary
Steve Underwood, Would you mind summarizing where/how T.38 functions, and maybe how it compares to the analog fax environment for the asterisk-users arhives? Seems to be some misunderstanding, and a lot of interest in handling faxes in various forms via asterisk. If some these approaches were summarized in one posting, a lot of us could reference it to remind us of limitations, current state,
2005 Jul 13
2
Faxing Suggestions
I wrote a week or two ago about a problem I was having with a TDM400P with 2 FXS. The problem was reliable fax transfers. There were often slow or had communication failures during transmission. Richard replied to my post and told me this is an ongoing problem that might have a solution in the weeks to come. However, I was wondering if anyone had any alternate solutions for faxing with an
2010 Jun 29
0
T.38 Peer Negotiation Fails
Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102 (yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of
2009 Jul 16
1
Mexican ITSP needed
Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:
2006 Oct 14
1
Codec swap (reinvite)
Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on