[Asterisk-Users] Problem with grandstream devices and DTMF signalling

ter a Playtones(), waiting for user entering an extension. I've tried many solutions, played around with all dtmf options...but no luck... Note that after the first digit (which get 'lost'), others get processed successfully. Cheers, Simone. -- _______

Re: [Asterisk-Users] Re: Send DTMF tone Like 'C' on connected call

You could use D's option of the dial() command. Something like Dial(SIP/xx,,D(C)). But check the documentation. Cheers, Simone. Antonio Rabena ha scritto: Hi, How can i send dtmf tone upon connection? ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] Problem with grandstream devices and DTMF signalling [RESOLVED]

Resolved with the help of a gentleman in chat :) BTW, i was lacking an Answer() in the dialplan. Cheers, Simone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options vi

Re: [Asterisk-Users] New $89 VOIP phone

That's not true: with some equipment you can use VLAN tagging to separate VLANs. This allows to have multiple vlan's running on the same wire. Cheers, Simone. Richard Cook ha scritto: I think the concept behind that is to have your voice on a separate VLAN then your data. In this case, a separa

Re: [Asterisk-Users] New $89 VOIP phone

Surely, 802.1Q wasn't designed with security in mind...change tagging, change vlan... Cheers, Simone. Steve Szmidt ha scritto: Thus the Virtual part of VLAN. Though it's still a very good idea, from a security standpoint, to keep them apart. You do not want to have your LAN owned because of the

Re: [Asterisk-Users] New $89 VOIP phone

I don't really think so. They only can tag traffic they generate, AFAIK. My ATA-186 from cisco has that functionality, but never tried myself. Peter Svensson ha scritto: I have not used any ip-phones with vlan support but several switches. They all allow you to confgiure which tags are accepted o

[Asterisk-Users] Problems with DTMF

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's pos

[Asterisk-Users] SpanDSP

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.

Re: [Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?

Hello All, Currently my setup uses Xlite and Asterisk and i found that all the RTP voice packets are transfered via the asterisk server from one xlite to another. Is there any possibility that we can make all the RTP Packets to be transfered directly between the two clients once the connection is e

Re: [Asterisk-Users] Asterisk as SMS Service Center

I've found app_sms which is supposed to do that. However, I never managed to get it work. Every phone I tried refuses to communicate with asterisk. This was my (very basic)config: exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,SMS(test,as) exten => 1,4,HangUp This is supposed to answer th

Re: [Asterisk-Users] Asterisk as SMS Service Center

Tried, doesn't work. And onestly, I've not catched this. Where's the difference? Cheers, Simone. administrator tootai ha scritto: Simone Ricci a écrit : I've found app_sms which is supposed to do that. However, I never managed to get it work. Every phone I tried refuses

Re: [Asterisk-Users] spandsp

FWI, I'm using it very successfully. Where's your problem? My system is running Fedora Core 2, libtiff and libtiff-devel as provided by distro (installed with apt-get, version 3.5.7). I've got to add some headers in /usr/include to get it work: tiffiop.h, tif_dir.h and ports.h Hope this helps.

Re: [Asterisk-Users] spandsp

William Glynn ha scritto: I'm seeing training errors that often stop fax machines from even starting delivery, lots of errors from libtiff regarding unexpected line lengths, and garbled data even if RxFax thinks it's okay. Strange. Which version of libtiff are you using? However, I've gotten TxFax

Re: [Asterisk-Users] spandsp

administrator tootai ha scritto: > > From where you got port.h? I install tiff-3.5.7 (I'm running an RH73 > with those rpms installed) and didn't manage to create this file. > tiffiop.h and tif_dir.h are parts of tar.gz package. I had to modify > tiffiop.h and replace port.h with tiffcomp.h but I'

Re: [Asterisk-Users] spandsp

Jon Bebeau ha scritto: 1) Where did you get the C headers from (what dir). tiffiop.h and tif_dir.h somewhere in the libtiff's 3.5.7 source .tar.gz port.h with google's help. I can send my version to you if you wish. It worked for me. 2) What HW are you using (processor, motherboard) Intel(R) Cele

Re: [Asterisk-Users] spandsp

William Glynn ha scritto: I'm using Gentoo's tiff-3.5.7-r1 package, which appears to be built from virgin (unpatched) 3.5.7 sources. So it must be ok. Strange, again. I've not tried loopback, but my incoming/outgoing faxing works like a charm, via SIP. I don't have any pstn hardware, my pstn inte

Re: [Asterisk-Users] spandsp

administrator tootai ha scritto: I just finish installation and give it a try on reception: same problems than related by William. Seems that around 20 first kb of data's are approximatively ok and the garbage start (unexpected line lengths) [ ... ] I use an X100P. SMP (2 celeron 400) 440 Mb RAM

[Asterisk-Users] Multi-bitrate codecs

Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... Cheers, Simone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http:

[Asterisk-Users] HFC-S in NT mode, wiring?

I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)?

[Asterisk-Users] Some values ignored when using static realtime

Hi, I'm experiencing a strange issue with static realtime. Seems that some values belonging to 'general' category (like, for example, rtptimeout, rtpholdtimeout, realm) are ignored. Running asterisk 1.2.1, I've tried both res_odbc and res_mysql (that one from asterisk-addons tarball) without luck.

[Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it

[Asterisk-Users] SIP Channel and jitter buffer

Hi, what's the current status of jb implementation in chan_sip? Are there any patches out there available to be applied to the brand new 1.2-release? Cheers, Simone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing lis

[Asterisk-Users] [OT?]SCCP image for cisco 7905g

Hi, I recently purchased a brand new 7905g with his SIP firmware (licensed). Now, I want to play a little with chan-sccp, but I'm unable to find the appropriate firmware for my phone. I know that I must get it directly from cisco, but before purchasing it will be very good for me to try a bit; does

Re: [asterisk-users] SER with multiple asterisk deployment

Adi Simon ha scritto: > Hi, > > Did anyone actually manage setting up a single SER with multiple > Asterisk boxes? > I particulary have a problem of keeping the session alive and by that I > mean directing > all the following sip messages to the same asterisk box the first signal > was sent (rand