search for: weppler

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.

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/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

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

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,

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

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

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

>> 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

Hallo.. if you have one more for me...thanky you.... Stephan -----Urspr?ngliche Nachricht----- Von: WipeOut . [mailto:wipeout@linuxmail.org] Gesendet: Donnerstag, 10. April 2003 19:28 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Asterisk-Redhat 9 install guide. Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

Let post this question.. Because I must be real slow... The following is my config on this... group=1 context=default signalling=fxs_ks channel => 1 context=local signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes

On the cheap side, the ITX or even MicroATX machines work great. These are commodity items, so they tend to be far less expensive than custom solutions. Various manufacturers, but we've had very good success with any of the AOpen MicroATX boards and their slimline MicroATX case: Aluminum: http://usa.aopen.com/products/housing/A340-series.htm Steel:

...t. I just 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 ex...

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

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

...ossible 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. >...