similar to: - SOLVED - Trouble getting SMS working

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 =

Hello, I tried to send sms for local extensions and i observed that file is created but sms isn't delivered yet. Can someone help me with this thing? rr:/var/spool/asterisk/sms/mttx # cat ../../outgoing/smsq.mttx.0.1322430026-20217.1 Channel: Local/1010 Callerid: SMS <1010> Application: SMS Data: 0,s MaxRetries: 0 RetryTime: 30 WaitTime: 10 rr:/var/spool/asterisk/sms/mttx # ls -la

Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as "This phone system will be stopped in 5mn for maintenance" to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short

Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---<Cut Here>--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22

Good afternoon, I am triying to send an sms through a gateway gsm (stargate) that is connected to a ZAP/g1 card on my asterisk. But I get this message : -- Attempting call on ZAP/g1 for application SMS(0) (Retry 9) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/1, span 1 got hangup -- Channel 0/1, span 1 received AOC-E charging 0 units > Channel Zap/1-1 was

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 -

I've been googling and reading a lot, but I'm not getting any closer to getting an SMS sent via Asterisk. Prior to switching to asterisk, I used sms_client on an ISDN line to dial one of two Swisscom SMS centers: 0900900941 or 0794998990. My dialplan looks like this: exten => 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1) exten => 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1) ;

Hello all, After reading through the Wiki and archives, I decided to take a stab at installing * on Solaris 9 SPARC. I checked it out via CVS last night as well as about an hour ago, and have run into a compile problem that I can't quite figure out. After running into some unlisted dependencies, such as popt, things are almost compiling. Right now the make bombs when trying to find setenv

Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 & ZAP/2). I've not had luck with either of my lines, after issuing the command "smsq --motx-channel=ZAP/1/1709400X 00000 register". I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for

Hi, I worked with Asterisk 0.7 without problems until I tryed to load H323. I downloaded the last version and after some try I compile it. I followed the description in /asterisk/channels/h323/Readme and the compilation of this part was good. But the new compilation of Asterisk was impossible (problem with chan_h323.so). I search info with Google and I read that the problem could be with the

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

Hi all, I?m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I?m on Spain, and I don?t know this can be a problem (with the operators...) I have Elastix 1.3.2 and I have seen this url: http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html I have tried the smsq command but I can get it work, (as I say I?m a begginer and I don?t understand several

> In case I wasn't clear: I don't think you have to modify the command > at all. Just create a template that uses the command as it currently > works. I thought we needed a new template only for the second log file? To generate the first log file using the existing $log command, I have introduced another $log command in query template that looks like:

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

All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same => n,SMS(hello,a,17654307001,"hello nick") - nick

How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? Yuan Liu

Hi, i've been trying to get fixed line sms working for some time now. Can anybody tell me, if he is actualy using this with asterisk in germany? I have followed the instructions found on voip-info. I was successful a couple of years ago with asterisk 1.0.7 and an normal telekom isdn line. Now i want fixed line sms over an Dokom PRI with Asterisk 1.2.9. Here in Germany the Materna

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE