similar to: Working Wellgate *SIP* 38xx/35xx hardware anyone?

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel

Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty

Recently, there're some posts about registration problems with this gateway. These are my observations using the latest firmware version 107a. a) The gateway will register all 4 ports if you're not using password. b) If using password, the gateway will only register the 1st port correctly. For the record, the 2-port 3502 doesn't exhibit this behavior. FYI.

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

Ok this is a simple question... What has been your experience with the WellTech 38xx series (I'm looking specifically at the 3802) VoIP gateway? I'm looking for a good (and hopefully not too expensive) VoIP/T.38 gateway for my office. Asterisk intergration is not a major factor at this time but may be later on. How well does it work? Is Echo a problem? Do the T.38 capablities

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =

Hi again, Kind of sheepish about asking for help, as I have only spent a day banging my head off this... I got my new Welltech 3701a, 1FXS,1FXO gateway. I flashed it with what is seemingly the appropriate firmware (SIP V1.04). This seems to have gone ok, and it is now registering both ports ok with asterisk. For 1 minute I thought I was home free and and everything was just going to work

Hi there I have some problems in setting up a connection between * and a Wellgate 1501 (1FXS). Either I can dial from * to my Wellgate or I can dial from the Wellgate to *. On the Wellgate you have to setup both "Line number" (which is the number it answers) and "Line account" (which is the account it registers as). Should I expect any problems if these are set the same?

Hi, I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64? I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection. Any ideas? Vahan

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

Anyone know how to set the wellgate unit so incoming calls pass on directly to asterisk? Right now incoming calls ring twice and I hear a recording saying enter the extension. If I go enter the extension it goes on to asterisk just fine. I just want the incoming call to go directly onto asterisk. Anyone found that out? Jerry

Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card.

What is the proper way of adding hints to multiple extensions? In my case extensions are the same as the sip usernames, while as per http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence exten => 1234,hint,SIP/1234 works, exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use ${EXTEN} here... Any hints? Vahan -------------- next part -------------- A

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540