Re: [asterisk-users] ATA FXO

Joseph, You may want to try RPA-2E1S1O from www.broad-tel.com from China. It provides real FXO port that registers with Asterisk. David On Sat, Dec 12, 2009 at 1:37 AM, Joseph wrote: > I'm looking for a reliable ATA FXO/FXS adapter. > > Linksys 3102 - a lot of echo problem + two of them died wit

Re: [asterisk-users] My AsteriskNo unable to registration

nning on host "127.0.0.1" and accepting > TCP/IP connections on port 5432? > > > Change in your sip.conf put : > > type=friend > > failed for 'xxx.xxx.xxx.xxx' - Peer is not supposed to register > > > Tellme something. > > Guillermo > > El Miérco

Re: [asterisk-users] My AsteriskNo unable to registration

Hello, could you please advise .. where can I find the trace of asterisk? do you mean log file? Thanks & Regards Bie - Original Message - From: "Guillermo Rodriguez" <[EMAIL PROTECTED]> To: "Newbie" <[EMAIL PROTECTED]>; "Asterisk Users Mailin

[asterisk-users] SPA-3102 Registration Failed .. need advise

Dear Expert, I am stuck when trying to register SPA-3102 on AsteriskNow .. could any body please advise .. where can I find the article for doing this? .. I googled but got nothing.. Regards bie ___ --Bandwidth and Colocation Provided by http://www.

Re: [asterisk-users] Registration state: Failed

Hi, I am using the OS which bundled with AsteriskNow - Original Message - From: Vivek Shrivastava To: Newbie Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 12:25 PM Subject: Re: [asterisk-users] Registration state: Failed

Re: [asterisk-users] Registration state: Failed

;172.16.1.169' - Device does not match ACL any idea or clue? Thanks a lot in advance Regards Winanjaya - Original Message - From: Vivek Shrivastava To: Newbie Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:50 AM Subj

Re: [asterisk-users] Registration state: Failed

Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - From: Vivek Shrivastava To: Newbie ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December

[asterisk-users] Registration state: Failed

Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 -> my extension 998 -> I configured as Line 1 in SPA-3102 999 -> I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the

[asterisk-users] My AsteriskNo unable to registration

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (bec

Re: [asterisk-users] # to transfer calls

ng blind transfer. I heard the annoucement "transferred" when '#' was pressed. Thanks. David On 9/24/07, Atis Lezdins <[EMAIL PROTECTED]> wrote: > > On Monday 24 September 2007 10:21:44 VoIP Newbie wrote: > > I wonder why my call was transferred when I pres

[asterisk-users] # to transfer calls

Hi all, I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation

Re: [Asterisk-Users] Use Grandstream ATA as trunk

You need a FXS to FXO converter between ATA and the GSM box. You can get one from www.broad-tel.com On 1/13/06, Ronald Voermans <[EMAIL PROTECTED]> wrote: Hi All,   I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has

[Asterisk-Users] DTMF dialing

Hi all,   I am trying to get DTMF digits from X-pro, through a grandstream ATA, to a FXS to FXO converter for outgoing PSTN calls. I could hear second dial-tone from the phone line connecting to the converter. However, no PSTN dialing occured after DTMF digits was sent from X-pro. I tried while X-p

Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount

i don't think this is a good idea in case someone is using your account and you can never know... On 1/2/06, Leif Neland <[EMAIL PROTECTED]> wrote: Original Message From: "Andreas Koch" < [EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"

[Asterisk-Users] USB phone

HI all,   I am wondering if asterisk supports USB phones.   Thanks. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-

Re: [Asterisk-Users] Asterisk & FXO & Panasonic PBX

There are 4 options for your consideration:   1. use 2 x 1-port FXO gateway 2. use 2-port FXS gateway with FXS to FXO converter 3. use a 4-port FXO gateway. 4. use 2 x X100P cards   You can get them from www.broad-tel.com  On 12/21/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: I'm looking for a r

Re: [Asterisk-Users] Affordable IP Phones for Asterisk

you want something really cheap. you got to visit www.broad-tel.com. It is even offering a WiFi phone at US$125 for its existing clients.  On 12/20/05, Dakota <[EMAIL PROTECTED]> wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer

Re: [Asterisk-Users] Re: T.38 ATA

PA-122TI from www.broad-tel.com supports T.38 and Fax pass-thru. On 9/15/05, Rosario Pingaro <[EMAIL PROTECTED]> wrote: about spa-2100, the t38 stream is on UDPTL and so asterisk passthroughdoesn't work.- Original Message - From: "Nenad Radosavljevic" <[EMAIL PROTECTED]>To:

Re: [Asterisk-Users] Zap Channel

Alex,   Context solved my problem. Thank you so much.   Happy David.   On 9/13/05, Alex Ongena <[EMAIL PROTECTED]> wrote: On Mon, 2005-09-12 at 23:34 +0800, VoIP Newbie wrote:> Below are what I have in extension.conf. Is this the complete file ?>> exten =&g

Re: [Asterisk-Users] Zap Channel

x27;t know how to differentiate calls from a particular channel. 2. I don't know how to make a channel no Answer.   Please advise and help.   Many Thanks,   David  On 9/12/05, Alex Ongena <[EMAIL PROTECTED]> wrote: Euh, what is your extensionf.conf part that answers it ?On Mon, 2005-09-12

[Asterisk-Users] Zap Channel

Hi all,   How can I make a X100P ZAP channel not answering to any incoming calls?   Thanks.   Newbie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] PLEASE REPLY, are you using an X101P

I bought 3 from 3 different vendors. One of them has echo issue. Another one has an issue regarding PCI master abort. Only one really works fine for me. These 3 cards use AMBIENT chip but with different layouts and SLICs. On 8/4/05, Mark Burton <[EMAIL PROTECTED]> wrote: > X101P with Ambient md320

Re: [Asterisk-Users] 8 FXS in Asterisk Server

Get a 8-port FXS gateway from www.broad-tel.com. That is the single box you need. On 8/16/05, Roland Zagler <[EMAIL PROTECTED]> wrote: > Hello everyone, > > I want to build an Asterisk Box where i need 8 FXS interfaces > to connect 8 phones to. The problem is, that there is only one > PCI slot av

Re: [Asterisk-Users] v92 modems

I got one from www.broad-tel.com. It works fine. On 8/12/05, Douglas Logan <[EMAIL PROTECTED]> wrote: > Yes, but your results may vary. Apparently some people have problems > with "clone" cards (aka regular modems), dropping calls, and having > echos. (Then again some people have reported no probl

Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service

lves as different > manufacturers. I'm assuming that they have different firmware versions > from the OEM. > > Mark > > VoIP Newbie wrote: > > I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, > > the other one has echo. Both use MD3200 ch

Re: [Asterisk-Users] FXO port trhoug optimum voice VOIP service

I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath <[EMAIL PROTECTED]> wrote: > Carlos Trallero wrote: > > >Hello, > > > > I have asterisk running on Fedora Core 3 with a x

Re: [Asterisk-Users] FXO gateways

You may want to contact www.broad-tel.com/index_en.php. They offer a variety of FXO SIP gateways from 2-port to 16-port. On 8/8/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote: > Has anyone found a suitable but not exorbitant 4-6 port FXO => sip > gateway? I need something more compact than a ch

Re: [Asterisk-Users] Phone interface hardware

If it is expensive to get a separate LAN connection for analog phone adapters, you can get one with 2 ethernet port and 1 FXS port such that it can connect your PC and analog phone over a single cable to the network. It is not difficult to find such kind of analog adapters for around US$50 or lower

Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

Tim, Can I test it as well? Best Regards, Newbie On 8/2/05, Vlasis Hatzistavrou - asterisk mailing list account <[EMAIL PROTECTED]> wrote: > > > If anyone is interested I'm (slowly) developing a GPL'd Java applet that > works as an IAX softphone. > > I sho

[Asterisk-Users] Zaptel warning

NG[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' Any idea for this problem? Many thanks. Newbie ___

Re: Re[2]: [Asterisk-Users] DECT VoIP Gateway

http://www.broad-tel.com/products/wireless.php On 7/7/05, Ola Lidholm <[EMAIL PROTECTED]> wrote: > >> > -Original Message- > >> > From: [EMAIL PROTECTED] > >> > [mailto:[EMAIL PROTECTED] On Behalf Of > >> > IM.Nobody > >> > Sent: Wednesday, 6 July 2005 11:51 PM > >> > To: Asterisk-Users@li

Re: [Asterisk-Users] DECT VoIP Gateway

I wouldn't mind such a single message. It is really a new breed of product that is not known to most of us. Correct me if I am wrong. While there have been some discussions on HOP-ON's wifi phone of $39 that never came true, this may be a sound alternative for all of us. On 7/7/05, Terry H. Gilse

Re: [Asterisk-Users] DECT VoIP Gateway

Would it be a good replacement of expensive WiFi phones? How much is it?? On 7/6/05, IM.Nobody <[EMAIL PROTECTED]> wrote: > Hi all, > > Just want to share with all of you a new hot DECT VoIP gateway > available from www.broad-tel.com/index_en.php. > > The DECT VoIP gateway is capable of handling

Re: [Asterisk-Users] raising the sound volume on zap

txgain/rxgain in zapata.conf. On 7/3/05, wassim darwish <[EMAIL PROTECTED]> wrote: > i noticed that the sound volume of the zap(tdm400p) > was low ,so i tried to raise the sound volume but i > didnt know how please help me. > > __ > Do You Yahoo!?

Re: [Asterisk-Users] Calls authentication by IP address

r (in sip.conf). Make sure, they > don't have to register -> host=123.123.123.123 instead of > host=dynamic. > > Julian. > > On 7/5/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > > Hi all, > > > > Is there any AGI supported ca

[Asterisk-Users] Calls authentication by IP address

Hi all, Is there any AGI supported calls authenticated by IP address? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] wi-fi phone advice

The one that looks identical is selling at $180 from www.broad-tel.com/index_en.php On 7/1/05, Richard Malcolm-Smith <[EMAIL PROTECTED]> wrote: > If it does materialize, im up for 3 or 4 of them at that price. > > Huddleston, Robert wrote: > > Well poo - if I can use that word I'm one of thos

Re: [Asterisk-Users] Hop-On WIFI Phone MSRP $40

Well, I found a WiFI phone that looks identical to the Hop-On one. It is from www.broad-tel.com/index_en.php but is selling at $180/each for every 20 units. On 6/29/05, William Suffill <[EMAIL PROTECTED]> wrote: > Unfortunately no. Someone say the press release and bugged me about it > as well but

Re: RE : [Asterisk-Users] Ambient MD 3200 (X100P Clone)

My one from www.broad-tel.com works fine and is very cheap. On 7/1/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Yes, I have :-) > 3 of this cards running well on my personnal * > What price for your ? > > Best Regards, > Francois BERGERET, > France. > > > -Message d'origine- > De

Re: [Asterisk-Users] Portable USB headset for VoIP

The supplier is from www.broad-tel.com On 6/14/05, Jian Hong GUAN <[EMAIL PROTECTED]> wrote: > That interests me. Can you send me the informations about products and > suppliers? > Best regards, > --Hong > > ___ > Asterisk-Users mailing list > Asterisk

Re: [Asterisk-Users] Portable USB headset for VoIP

There is an USB softphone or MP3 softphone that you may find useful. The USB softphone in size of USB flash disk comes with built-in sound drive. It can embed a softphone such that it is portable anywhere even an computer does not equip with a sound card. It is also a USB flash disk that can be use

Re: [Asterisk-Users] FXO Gateway recommendation

Please visit www.broad-tel.com for details. On 6/8/05, Wai-Sun Chia <[EMAIL PROTECTED]> wrote: > On 6/8/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > > My 4-port FXO is only $300. > > > Which product/model are you using then? > > /wai-sun > _

Re: [Asterisk-Users] FXO Gateway recommendation

My 4-port FXO is only $300. On 6/8/05, Adrian A <[EMAIL PROTECTED]> wrote: > >From your experience, would you recommend purchasing 8 Sipura 3000 1 > port FXO gateways or 1 Audiocodes 8 port FXO gateway? > The way I see it, the advantage of going to the Sipura solution is > that it is more scalable

[Asterisk-Users] Duplicate Calls

I am using Asterisk CVS-HEAD-06/02/05-19:37:27. It seems that every call I made was duplicate. Jun 8 00:11:30 DEBUG[21733]: chan_h323.c:411 oh323_call: Placing outgoing call to 87874586, 101 -- Called 87874586 Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using address 10.17.

Re: [Asterisk-Users] 4+ Port FXS Analog Device

; behind NAT and potentially, the asterisk server will be behind > another NAT. > > Thanks, > - Waldo > > On Jun 2, 2005, at 3:23 AM, VoIP Newbie wrote: > > > Another alternative is to use H.323 FXS IAD in combination of H.323 > > channels. I bought a 4-port IAD

Re: [Asterisk-Users] 4+ Port FXS Analog Device

Another alternative is to use H.323 FXS IAD in combination of H.323 channels. I bought a 4-port IAD of US$50 per port. It works for me!!! Let me know if you will be interested in the product. On 6/2/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > I'm looking for an inexpensive way to connect 20

[Asterisk-Users] RTP Read too short

Hi, I am using chan_h323 from CVS. An incoming call from H323 phone caused the following error: Jun 2 20:07:08 WARNING[1166]: rtp.c:457 ast_rtp_read: RTP Read too short Any idea for the above message?? Thanks. Newbie ___ Asterisk-Users mailing list

Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software

Does it support pre-paid billing? On 5/30/05, Darren Wiebe <[EMAIL PROTECTED]> wrote: > El Flynn wrote: > > > Darren Wiebe wrote: > > > >> Good Day, > >> I'm finally getting around to officially announcing ASTPP. For the last > >> 6 months, I've been working on converting ASTCC into a decent bil

[Asterisk-Users] Re: SIP and FastStart

Can anyone give me a big here? On 5/13/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must > be done to make FastStart work with SIP phones. Thanks. > > On 5/12/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: &g

[Asterisk-Users] Re: Always Ringing

Can anyone give me a big hand here?? On 5/16/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > Hi all, > > I am using chan_h323 from Asterisk CVS to interconnect with GNUGK > v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on > Asterisk. However, I only heard ring

Re: [Asterisk-Users] asterisk-oh323 build problems

Read README file first. You will get a clue. On 5/19/05, FaberK <[EMAIL PROTECTED]> wrote: > Hello Guys, > first of all, I'm very new with asterisk. > I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 > Now I'm trying with asterisk-oh323 > I've already installed pwlib, oh

Re: [Asterisk-Users] Asterisk with modem

You are wasting your time while you can get an OEM X100P for a few dollars. Check it out at eBay or www.broad-tel.com. On 5/18/05, ALIF Mohssine <[EMAIL PROTECTED]> wrote: > Hello Dave, > Could I know why please ?? Thanks ! > > Dave Cotton <[EMAIL PROTECTED]> a écrit: > On Wed, 2005-05-18 at 11:2

[Asterisk-Users] Asterisk Aborted

Below happened when I am using Asterisk-oh323 0.7.1 with FastStart enabled. I made calls from H323 EP to SIP EP. After a long ringing at the orginating H323 EP, * was aborted as follows. Any help??? *CLI> 2:40.912Housekeeper PWLib Assertion fail: Function ::close failed, fi

Re: [Asterisk-Users] h323

CLI> show module and look for chan_oh323.so If oh323 is loaded, "oh323 show conf" will provide more useful info. On 5/17/05, Micko <[EMAIL PROTECTED]> wrote: > Hello! > > How can I check if oh323 is loaded and working? Is there a quick test for > this? > > Thank you. > > Micko > ___

[Asterisk-Users] Always Ringing

Hi all, I am using chan_h323 from Asterisk CVS to interconnect with GNUGK v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on Asterisk. However, I only heard ringing when the call was answered on SIP side. Below is the debug from chan_h323. Any help is welcome. Thanks. *CLI> == Ne

[Asterisk-Users] Re: SIP and FastStart

I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must be done to make FastStart work with SIP phones. Thanks. On 5/12/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > Hi all, > > When I enabled faststart in oh323.conf, calls from H323 endpoint to > SIP phones could

[Asterisk-Users] SIP and FastStart

Hi all, When I enabled faststart in oh323.conf, calls from H323 endpoint to SIP phones could not complete. The originating phone kept ringing when calls were answered by SIP phones. fastStart=yes h245Tunnelling =yes h245inSetup=yes Please can you advise. Many Thanks. __

[Asterisk-Users] Re: Connecting to provider

Sorry, I just fixed it by myslef. It is an issue of incompatible codec. I am wondering why option "t" in dial() is not able to make it work. Any advice??? Many Thanks. On 5/6/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > Hi all, > > I could register * to a provider.

[Asterisk-Users] Connecting to provider

Hi all, I could register * to a provider. However, I failed to make outgoing calls through the provider. Please help and advise how to get it work. m2*CLI> sip show registry HostUsername Refresh State sip_proxy:5060 abc105 Registered

Re: [Asterisk-Users] Re: V92 modem with asterisk

You can get one at around US$6 from www.broad-tel.com, including installation instruction. On 4/5/05, Tore Hansen <[EMAIL PROTECTED]> wrote: > You do need a proper FXO card to connect your POTS line > However, that need not be expensive. A suitable card is > available by mail order in the U.S. fro

Re: [Asterisk-Users] wcfxo problem

My one from www.broad-tel.com works perfectly. On 4/11/05, Sahil Gupta <[EMAIL PROTECTED]> wrote: > I'm having similar issues using an X100P Ambient Chipset Clone Card > any ideas? > > Regards, > > > Sahil Gupta > VoiceValley > > On Mon, 11 Apr 2005, Dave Weis wrote: > > > > > I've got a

Re: [Asterisk-Users] Re: Ring Twice

ing Dial("SIP/1000-a071", "OH323/001234|60") in new stack -- H.323 call to 001234 with codec(s) g729 -- Called 001234 -- OH323/L17774 is ringing -- OH323/L17774 answered SIP/1000-a071 -- Hungup 'OH323/L17774' Thanks for help. Newbie On Apr 8, 2005

[Asterisk-Users] Re: Ring Twice

Hi all, Did I make my issue clear? Can any one give me a big hand? Many thanks. Newbie On Apr 5, 2005 12:59 AM, VoIP Newbie <[EMAIL PROTECTED]> wrote: > Hi all, > > When I made calls from SIP phones through a analog PSTN gateway to > PSTN phones, I could hear rings twic

[Asterisk-Users] Ring Twice

ring like we make normal analog PSTN call from a normal PSTN phone? Many Thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Registration to multiple GKs

Hi all, How can I configure chan_h323 or chan_oh323 to register to multiple GK and route calls in-between? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] H323 gateway thru NAT

But success depends also on whether the router can do port > forwarding and whether the H323 Gateway supports NAT. > > This is possible with Quintum for instance with some port forwarding rules on > router level. > > Selon VoIP Newbie <[EMAIL PROTECTED]>: > > > Hi

[Asterisk-Users] H323 gateway thru NAT

ve the above? Please help and advise. Many Thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mail

RE: [Asterisk-Users] RE: Inexpensive Analog Ports

Mon, 2004-01-05 at 01:38, Asterisk Newbie wrote: > Does anyone know of any inexpensive alternatives to the four port > analog module offered by Digium ($305) what work seamlessly with > asterisk? > > > > Thanks -- Go to http://www.digium.com/index.php?menu=document

[Asterisk-Users] RE: Inexpensive Analog Ports

Does anyone know of any inexpensive alternatives to the four port analog module offered by Digium ($305) what work seamlessly with asterisk?   Thanks