similar to: Asterisk 1.4.17 and RXFAX via T38

Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones.

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, I think I asked this previously but don't remember seeing an answer... Yes, you can "tickle" an SPA94x or 962 and have it fetch a config from a TFTP server... But is there no way to simply "push" a couple of lines of XML config to it directly via an HTTP POST (sans TFTP server)? Thanks, -Philip

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

Hi all, Here's our setup: Asterisk 1.4.18 Agx-ast-addons 1.4.5 Problem: When accepting a fax, the fax itself comes through just fine, and it does successfully create a tiff file. However, the dialplan should be executing a system command right after that completes, but isn't due to hanging up early. I'm getting a cause 16 hangup, which I believe is a "Normal Hangup", but

Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [fax at phones:1] Set("Zap/2-1", "FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif") in new stack -- Executing [fax at phones:2] RxFAX("Zap/2-1", "/var/spool/asterisk-fax/1207322398.0.tif") in new st ack [Apr 4

On 1.6.1-beta4: Trying to receive faxes over a pstn line. extensions.conf: [incoming-pstn-line] exten => fax,1,NoOp(Fax Detected) exten => fax,2,GoTo(incoming-fax,s,1) exten => fax,n,Hangup() ............ [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, anyone else seen RX/TXFax crashing Asterisk on latest Asterisk SVN? I've now seen it on two machines I tried to set up - one it seems because the tiff file was malformed, but the other is doing: tiff -> tx fax -> zaptel -> pstn -> ddi -> zaptel -> rx fax -> tiff The above crashes every time. If no one else has

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

Hi, I installed project IGI with the CD , everything seems to work but there are a small mouse bug : i can't turn more than 180? :( I think this bug can be resolve if I "fix" the mouse somewhere on the screen but don't work in winecfg. OS:Ubuntu 7.10 Wine Version:0.9.55 card driver:nvidia-glx-new with the repository

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hi All, I've been knee deep in T38 faxing for a couple of weeks now, trying to find a version of Asterisk that would pass through T38 with an Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x through 1.6.1.10. Tested 6 different versions. Either it just would not work or fail back to G.711, or re-invite with wrong T38FaxMaxDatagram sizes, faxes would work one-way and not