similar to: Asterisk as a PSTN gateway for SER

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 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 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,

Sorry for the simple question. I'm trying to match "sipprovider.nocredit" but the following doesn't execute NoOp (it runs "context" but not "context-custom"). What am I doing wrong? [context] include => context-custom exten => _.,1,Set(GROUP()=1) exten => _.,n,Goto(destcontext,${EXTEN},1) [context-custom] exten => sipprovider.nocredit,1,NoOp(No

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

What we would like to see happen or emulated is that if someone makes a call via our SIP provider to a PSTN number that is actually busy that we get an actual BUSY tone at the telephone. In our test case this is a PAP2-NA SIP device It would appear that when we call the far end (PSTN phone number) that is busy we do not get any busy indication at the user end (originating telephone on our

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

I am trying to set up an IAX2 trunk between two servers. Server A has the following in iax.conf: [general] ... [ServerB] type=friend trunk=yes host=dynamic secret=myPassword context=myContext Server B has the following in extensions.conf: [outgoing] exten=>_40X,1,Dial,IAX2/ServerB:myPassword@x.x.x.x/${EXTEN} I am using bmtools to monitor the bandwidth usage, and I am not seeing a difference.

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

I recently switched from Mandrake to Redhat and I noticed that vmail.cgi does not work with the default apache installation that comes with Redhat. Here is what I get in my error logs: [Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism enabled (wrapper: /usr/sbin/suexec) [Thu Dec 04 11:59:58 2003] [notice] Digest: generating secret for digest authentication ... [Thu Dec 04 11:59:58 2003]

Dear ALL: My scenario lists below: Assume: UA1 with sip id "1011" And dial number to PSTN is "0939749xxx" There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==>

We're having a problem with Asterisk when we try to pass a call off to a Lucent PSTN using SIP. This behavior does not exist with SER: With Asterisk An ISDN call is started, at the T1 level we receive ?call proceeding? and immediately we receive a ?Call in Progress? just like the far end party has answered. With SER An ISDN call is started, at the T1 level we receive ?call proceeding?

Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten => s,1,Dial(ZAP/1/6135551111,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using