similar to: Asteriks Compile error

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

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 - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

At the last step of building llvm for arm target, I am unable to build llvm-gcc. I am trying the follwing options:   $ ../llvm-gcc/configure --target=arm-linux --enable-pic --program-prefix=llvm- --prefix=/llvm/install --disable-multilib --disable-optimized --enable-bindings =none --enable-llvm=$PWD/../llvm-2.8 --enable-languages=c,c++ --disable-bootstr ap   Then I get the following error when I

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

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

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

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 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, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? TIA. -- Bill in Denver

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

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) ;

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one