similar to: Help needed with IAX behind NAT

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

Does anyone know if this unit works with Asterisk? Thx. B.

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

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, I am trying to compile Asterisk under Yellow Dog 3.0 distributionn. I am getting an error gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm /usr/bin/ld: cannot find -lgsm May be I need packages that my distribution does not include? What do I need to download to get it compiled? Thanks Serge _________________________________________________________________ The new MSN 8: smart spam

Hi I am using inAccess channel driver. Compiled, installed. This is what I get when I am trying to start * --------------------------------------- [chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource): libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file or directory WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so

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 trying Manager interface for originate a call. This is what I get --------------- Action: Originate Exten: 555 CallerID: test <6656> Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed ---------------- What do I do wrong? Thank you Serge _________________________________________________________________ MSN Premium with Virus Guard

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

Hi, I updated kernel for RH8.0 and updated * from cvs. After that zaptel compile exits with error. Is it because of new kernel or zaptel source code change? In any case could somebody help me to fix this problem? Thanks, Serge ---------------------------------------- /usr/src/linux-2.4/include/linux/module.h:196: warning: parameter names (without types) in function declaration In file included

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

I'm new to asterisk and need some help. I have two computers. Each computer has a Wildcard X100P and a NIC. I have an incoming telephone line going to the "line" connector, and an analog phone connected to the phone port. I have everything installed, but I'm not sure what I'm supposed to do next. How do I go about calling one phone from the other?

Hi I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on another box behind of it and it seem to work fine for me. I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. Does anybody know of

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

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 am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using