similar to: Debugging DTMF

Hi All, If I had a large block of code, eg: [outgoing-pstn-gradwell] ; the caller ID convertion assumes that the last two digits of the callers id ; are mapped to the last two digits of the PSTN number. exten => _0.,1,ExecIF($["${RECORDOUTBOUND}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CA LLERID(num)}-${EXTEN}-${UNIQUEID}.WAV) exten =>

Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer

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Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility,

I'm getting this error message and I don't really understand what rsync is trying to tell me:- rsync: link_stat "/rdiffBackup/gradwell/Mail/." failed: No such file or directory (2) rsync error: some files could not be transferred (code 23) at main.c(977) [sender=2.6.9] Can anyone explain what it's saying please. /rdiffBackup/gradwell/Mail/ does exist and is

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]

Hello all. We have an asterisk system set up, and we are seeing a lot of multiple DTMF digits being read by asterisk. In digging through the archives the only answer I have seen is to put in the statement relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata devices, I have tried to put that statement in my sip.conf file to no avail. Any help would be appreciated as my end