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Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

What is the state of speex support in asterisk? I saw the codec seems to be there. Can speex be used on IAX2 links? Is there much work still to be done? many thanks, --J.

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This

Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

...st thought I'd mention it for you to possible look into. Thanks again for this great addition. Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Monday, April 14, 2003 7:23 AM To: 'asterisk-users@lists.digium.com' Cc: 'weppler@wwworks-inc.com' Subject: RE: [Asterisk-Users] RE: [Asterisk-Dev] Several patches, including recording and music -on-hold Hi Wade, Sorry for replying so late. I had been sucked into other tasks for a while and only now can catch up with the list. > When I dial my iaxtel number from my extension...

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

...look into. > > Thanks again for this great addition. > Ben > > > -----Original Message----- > From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] > Sent: Monday, April 14, 2003 7:23 AM > To: 'asterisk-users@lists.digium.com' > Cc: 'weppler@wwworks-inc.com' > Subject: RE: [Asterisk-Users] RE: [Asterisk-Dev] Several patches, > including recording and music -on-hold > > > Hi Wade, > > Sorry for replying so late. I had been sucked into other tasks for a > while and only now can catch up with the list. > > &...

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

...look into. > > Thanks again for this great addition. > Ben > > > -----Original Message----- > From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] > Sent: Monday, April 14, 2003 7:23 AM > To: 'asterisk-users@lists.digium.com' > Cc: 'weppler@wwworks-inc.com' > Subject: RE: [Asterisk-Users] RE: [Asterisk-Dev] Several patches, > including recording and music -on-hold > > > Hi Wade, > > Sorry for replying so late. I had been sucked into other tasks for a > while and only now can catch up with the list. > > &...

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hello! Several of my customers would like to add a backup to their Internet connection. ISDN is a good solution: decently fast for a dial-up-type connection, yet still faily affordable. While I was at it, I decided to look at a couple of more creative telephone service options to possibly improve their service or lower costs at the same time. These customers range from having just a

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

I was just trying to find a better way to transfer incoming calls to external phone numbers without tieing up my lines. The following has worked successfully for me and I just thought I'd post it so if someone was looking to do the same they could quite easily. The feature you need installed on your lines is called conference-drop-transfer or here in canada it's know as

>> We're having a problem with transfering calls. Our channels are not the >> same as the extensions. We use words instead of numbers. So our config >> looks like this: >> >> SIP/HRUTTER, 1, "81101 Hildegard" >> SIP/JFOLEY-GS, 2, "81103 Jerry" >> >> Consequently when I drag and drop to transfer a call

Just thought I'd share my success with chan_h323 and our Symbol NetVision phone (4046-100-US). Voice quality is excellent, and setup was trivial. The new NetVision firmware (4.21) is much better than the 3.x stuff. It gives the phone a whole new look and feel. The hardest (and longest) part was getting OpenH323 compiled. After that, H.323 ran out of the box. I simply uncommented