search for: fastbusi

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual Band Analoog FXO) working with Asterisk. I had a working FXO configuration to a analog port of a small home 1/4 ISDN pbx. I used this same configuration to connect a GSM Gateway that is supposed to be connected to the external(FXO) analog port of a pbx. I can get my configuration to dial the mobile number via the gateway, but

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I have some strange behavior happening with call flow when analog line errors are encountered. This may be due to the way that the X100P detects "busy" signals, or it may be something in the software. Could someone with more in-depth knowledge make a comment on the items below? My dialing logic says "dial local area code numbers out of the analog line, and if the analog line

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {

Hi, I wish to connect several ATA186 Phones to each other, to iconnecthere and to the PSTN using asterisk. Please tell the appropriate settings for firewall (ports to open etc.) sip.conf and extensions.conf(part relevant to iconnect). Also I would be glad to get a working example of your ATA186 configuration. I tried searching the mailing lists and several sites but did not find an answer.

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I'm using Asterisk 1.09 with bristuff 0.2.0-RC8n, one BRI line and a Sipura SPA-2000. I have a HP LaserJet 3330mfp all-in-one. I can receive faxes but not send them. The faxes start whistling to each other but the transmission is stopped with a "communication error" To receive a fax I have this in my dialplan: exten => 0000000000,1,Ringing exten => 0000000000,2,zapEC(off)

I'm getting a random fast busy signal after two rings on an incoming ZAP channel through an FXO interface on a TDM400P. I have a NoOp in the dial plan showing the callerID. When the random fastbusy happens, no CallerID is sent. When it works, the CallerID information is sent. Is this a card issue or a telco issue? I have a very upset client and need to figure out how to resolve this.

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

I am new to Asterisk (1.4.5), and I am trying to get chan_mobile working. My intention is to use it as a cheap GSM gateway. In the dialplan I configured that all mobile numbers should go thru the mobile channel. The current situation is that I can setup the call via the mobile channel (bluetooth), but sound is still thru the bluetooth attached phone's speaker and microphone. I searched

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,