similar to: problems with sending fax from SIP channels

1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong

Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in between....if I'm wrong....eh...oh well.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

I am having an odd problem with a linksys pap2 ata and asterisk... Asterisk won't detect digits from it until I issue a 'sip debug'. As soon as I turn on sip debugging, everything works perfectly (classic heisenbug)! Asterisk is latest Debian 'etch' packaged 1.2.13. sip.conf looks like: [mc_ext01] type=friend secret=ext01 context=mc_ata_in host=dynamic dtmfmode=rfc2833

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect

The best I can do so far (which appears to be a bit of a hack) is (<:0>S0), which says to add a '0' to the start of the string and dial immediately. This gives asterisk an extension dialled of '0', which isn't the 's' that i'd hoped for, but is a good start! (S0) by itself doesn't work, nor does (<:>S0). Any other suggestions? Thanks James >

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

I just registered ID 0004283 at http://bugs.digium.com for the problem described in subject (found when using a Linksys PAP2-NA). I don't know where the proper forum is to discuss, so I'm hoping anyone interested will read the bug and let me know your thoughts, either at bugs.digium.com, here, or by emailing me directly (or, please suggest another forum that is more appropriate). As

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

I've got a Linksys PAP2 on my Vonage account with unlimited usage, but my softphone-addon account only has 500 minutes. Anyone ever try to setup their * to mimick the Linksys PAP2 ? Any comments or suggestions on what problems I might run into if I try? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but when I tried sending the received fax file to a fax machine, I either get "line error" or just a blank page. Is anyone using Spandsp to send fax to fax machines on PSTN? I've run out of things to try now, and I'd really appreciate if

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Is there a way to configure the linksys pap2 to automatically connect to asterisk on lifting the handset (presumably into the 's' state)? Asterisk would then be listening for DTMF tones to figure out what to do rather than having to put a dial plan into each pap2. I think the pap2 is pretty much the same inside as a few of the sipura boxes so the same thing might work if anyone knows...

I am using my first PAP2 device from linksys. Used many polycom phones... I configured the PAP2 device with asterisk. I have the registration, thought I was good to go. Plugged in my Valcom 2924 public address analog connection, called the extension and I got busy... very strange I thought. I then looked at the status page of the PAP2 and it says the following Reg online and hook state OFF.

>Message: 7 >Date: Sun, 29 Oct 2006 22:00:22 +0100 >From: "Jose Limeres" <jlimeres@gmail.com> >Subject: [asterisk-users] Linksys PAP2: calling tone stops after 5 > tones >To: asterisk-users@lists.digium.com >Message-ID: > <2b3431b20610291300u420116e5scf9103d7dac54321@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed >

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to