similar to: fixlocalprefix error

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

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)

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

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:

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

I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock ======================================================== Make sure that you _dont_ have rtc support compiled into your kernel! INSTALL: make USE: make load REMOVE: make unload I interpreted this as

I want to set up a dial rule like this 9304752#w9#w+NXXXXXXXXX The point of this is. It will dial into a pbx with the account number 9304752, wait a second, dial 9 to get an outside line, wait a second for the outside line, and then dial the number to be called. When ever I save this in amp anything after the first # disappears. When I try to call out it doesn't do the adding of the

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

Apologies if this has already been posted to the list. I sent it about an hour ago and it hasn't shown up yet. I've changed the format to plain text in case that was the issue. I'm having a major issue whereby voice is going out fine but none is coming in. If I ring someone I will hear them answer and then nothing. They still can hear me. Looked at the logs and with my

What is the most efficient way to allow inbound callers to dial internal users yet restrict them from outbound PSTN calls? Today I have a basic greeting that after a welcome message allows inbound callers the ability to dial any of my users. However, it seems that since I transfer the inbound caller to a context that allows them the ability to call my internal users they have the same rights as

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,