similar to: Occasional SMS problem

Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read

Hi, I've been trying to setup SMS on asterisk - would be useful to have for things like server outages, email from important customers, etc. I can send SMS with no issues, although I have to send it over the Zap line.. none of the VOIP providers will route the call. It arrives on my mobile phone a couple of minutes later (usually.. had to wait half an hour for one). Incoming however just

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

I'm trying to send an sms: smsq --motx-channel=CAPI/g1/0809101000 0607396666 "X" It seems to try to do something, but FT aren't happy: -- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1) == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1) [Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: CAPI/ISDN4#02/0809101000-1 already

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

asterisk-16.8.0 Hi I've set up a callback script to retry a number if it's busy, but as I watch the console output asterisk seems to rush 3 or 4 calls at once before waiting the RetryTime of 20 seconds that I've set. The script: -----8<------ CALLERID=$1 EXTENSION=$2 TEMP=`mktemp /tmp/call-XXXXXX`.call cat <<EOF > $TEMP Channel: IAX2/account at

I am working on heavily optimising unusually static C++ code, and have encountered a situation where I basically want an optimiser that would speculatively unroll a loop to see if the first round of the loop could be optimised further. (I happen to know that it is possible.) The previous optimisations that produce the loop in the first place already do a magical job (relying heavily on constant

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hi, I have been trying to get my asterisk box to send SMS's to my Panasonic dect phone via a Linksys pap2. I believe I have the message centers setup correctly between * and the phone. The pap2 is configured to only use G711a. The Asterisk version is 1.0.7. In my /etc/asterisk/extensions.conf I have [smsphone] exten = 199,1,Goto(smsmorx,${CALLERIDNUM},1) [smsmorx] exten =

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by

Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and

I can make calls over a SIP trunk as SIP/<trunk>/number I am trying to make calls over an extension thought using the same format SIP/4452/number - its not working. person says they can connect a software as extension 4452 and it works just fine. I have my register: register => 4452 at X.X.X.X/4452 [4452] type=friend username=4452 host=X.X.X.X allow=all dtmfmode=inband When I try to

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its

Hi all. I am searching for a way to send SMS via our E1 PRI line. We are in Portugal and I have seen some Internet/TV/Phone providers (ZON for those who know it) who install normal phones with SMS support in landline. So I just found a page from PT (Portugal Telecom) stating that the SMC number is either 12999 or 129990 (

Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3&SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3&SIP/test1 for 1000 at from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]:

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

I wonder if anyone has come across this odd behavour with a T1 PRI using NI2 signalling from a Nortel switch. Sometimes, when bringing up a PRI trunk, a channel gets into a state where asterisk can't request a channel, and gets reason 0, but the channel is not busy. The only thing so far that clears this state is to make an incoming call to the channel, which succeeds. After that, outgoing

I'm using Slackware 11. I unistalled the package that provides libtiff 3.8..... and installed the most current 3.7.... for lib tiff. I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them. created a simlink: ln -s asterisk-1.4.0-beta3 asterisk I've compiled spandsp from as follows cd /usr/src wget

Hello! I think I've found a bug in some (older) versions of rsync. The problem does not seem to have been reported at bugzilla. And of the 9 bugs found there by searching on "--delete-after" all have got the name wayned@samba.org attached. I ask you therefore to check if this problem is found in the newest version of rsync as well! I've read the warnings at rsync.samba.org