similar to: Unable to build smsq on beta6 and x86_64.

Hi all, I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32 machine, and can't find any problems. So, I decided to upgrade my home pbx. All went well until I tried using my S101 to talk to the IVR. Some times, the first one or two digits get through, but eventually a digit will get stuck, playing continuously until the call is terminated. I have confirmed this

Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling

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

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 -

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

As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I do seem to remember that in 1.2, it wasn't possible to send a message to ones-self. This bug fix apparently corrects that situation. Well, I guess it would, if only it

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, 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]

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

> > I now have another problem with sending messages back to the phone, > when I run: > smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud "test" > You need to run smsq as the Asterisk user, or else the file is created with permissions that Asterisk can't read and/or move. James

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) ;

Help, Any ideas ? I guess I missing something. make[1]: Entering directory `/usr/src/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-HEAD-12/27/04-21:28:39\" -DASTERISK_VERSION_NUM =999999 -DINSTALL_PREFIX=\"\"

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

http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta6.tar.gz http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta6.tar.gz.sig Unfortunately this still isn't rc1 yet, but the next one will be, I promise :) Basically all of the important changes are done now. The TODO list for rc1 is: - Test and fix dsync bugs - Finish director - Add man pages started by Pascal Volk. BTW. He asked for

http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta6.tar.gz http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta6.tar.gz.sig Unfortunately this still isn't rc1 yet, but the next one will be, I promise :) Basically all of the important changes are done now. The TODO list for rc1 is: - Test and fix dsync bugs - Finish director - Add man pages started by Pascal Volk. BTW. He asked for

I released "aoTuV" beta6 of OggVorbis encoder, library. ---------------------------------------- CHANGES (for general) from beta5.7 # Libvorbis aoTuV was unified with Xiph.Org's libvorbis1.3.2. # Pre(post)-echo of the specific pattern was improved. # The bit allotment when sound of the different type coexisted was improved. # The calculation method of dynamic thresholds

Fixes, mostly: * The login and master usernames were reversed when using master_user_separator (now the order is UW-IMAP compatible). * Killing dovecot master process now kills all IMAP and POP3 processes also. + -a parameter to dovecot prints now all settings that Dovecot uses. -n prints all settings that are different from defaults. + Added pop3_lock_session setting + %M modifier