similar to: ZAP divert problem

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

What is asterisk's behavior when faxdetect=outgoing in zapapata.conf? Does it turn off echo cancellation? Does it also change the priority to "fax" in the "outgoing context"? Thanks, Vieri

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

So I have an application where the users want to divert incoming calls on one analogue line out to another analogue line - both lines are supplied by BT and theres a TDM400 in the box. Call comes in, system Dial's the forwarding number and bridges the calls. Works fine. Until one (or both) leg hangs up. And the box never sees the hangup. So it sits there with both Zap channels open

Hi All, I am looking for a way to allow users to dial *21*, followed by a number and the pound key. Asterisk must then divert all incoming calls to the user's extension to the number given. I got it right halfway using a tip from http://ns1.jnetdns.de/jn/relaunch/asterisk/page6.html. Thanks to whomever is the contributor of the above tip. The problem is however, that the number to which

Hi, sorry for cross-mailing. Reply-to: set to freebsd-net. I have seen some discussion on freebsd-security etc. about some parts of the subject. I have seen older messages in archives. Regularly the same questions seem to come up. I have not found an all-including description of the answer to s.th. like: "Can anybody tell me the order packets get processed in kernel related to IPSec,

> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE

I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs). With Trixbox out of the mix and a regular phone connected I get the CID fine yet Trixbox shows 'unknown': dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'ringall' Here is my Zapata.conf if it helps: ############################# ; ; Zapata telephony

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

I am having a problem with the tranfer function since changing our t1 to PRI. I used to be able to answer a forwarded call to my cell phone, then transfer the call back using ## (we have it set to ## in features.conf). Since the update it does not work most of the time. At lunch today I had a call and could not get it to transfer. The logs show asterisk getting the DTMF #, then a bunch of

I have a issue trying to understand why Asterisk-PBX, when a SNOM (320 or 360) successfully redirects/diverts a call when it is a local extension, but fails when you enter external number. Both the local extension dial and external extension dial are within the same context [from-sip] and both phones are capable of making external calls. I have looked at the standard sites, but not

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

hi all, when i start * all zap channels get ring signal so i get a huge number of incoming dummy calls when starting *. i'm using TE105P with 4 TA750 full fxo with latest CVS HEAD: zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsks=49-72 span=4,0,0,esf,b8zs fxsks=73-96 loadzone=us defaultzone=us zapata.conf: [channels] context=incoming

I have a strange problem: if I put on hold an incoming call from my Digium TE110P, I can't resume it and the person at the phone continues to hear MOH until the line falls. My TE110P is connected with an italian E1 NT. If I put on hold a call on a SIP channel I can resume it without any problems. Is there someone that can help me? These are my configurations: zaptel.conf