Q: The system cannot detect the internal module.
A: It is usually because there is no power connector. If there is an internal module on the card, you need to plug the power plug from the PC chassis into the power interface on the right of the card.Q: no external host is detected on the simulated card.
Q: The audio of the pickup signal is inconsistent with that of the normal telephone call.
A: domestic users often report this problem. You need to modify the configuration file:
Change loadzone and defaultzont to CN in l/etc/zaptel. conf
L enable busy tone detection in/etc/asterisk/Zapata. conf: busydetect = Yes
You need to reload zaptel and asterisk, or restart the system. Q: When simulating an external card call, the call is occasionally interrupted.
A: This problem often occurs when the quality of the wire connection or line is poor in some places.
You need to modify/etc/modprobe. conf and change the line install wctdm to the following:
Install wctdm/sbin/modprobe -- ignore-install wctdm battdebounce = 128 &/sbin/ztcfg Q: The call is noisy and echo exists.
A:
1. Activate echo cancellation in the configuration file Zapata. conf. Set the following options to yes:
2 echocancel = Yes
2 echocancelwhenbridged = Yes
2 echotraining = Yes
2. Adjust the rxgain/txgain settings to achieve a good echo elimination effect, Zapata must be adjusted. the rxgain (input gain) and txgain (output gain) values in the conf configuration file must follow these steps:
1. Change the rxgain/txgain values to 1.0;
2. Use the channel on the asterisk card to make a call, that is, you need to establish a call through the card channel;
3. Run the ztmonitor command to check whether the channel receiving and pronunciation gains are in a relatively optimized range. (Note: The first parameter of the ztmonitor command is the channel number you are calling)
4. Adjust rxgain and txgain to an optimized value based on your actual situation. If the values of rxgain and txgain are too small, you can add the values of rxgain and txgain to the configuration file Zapata. conf.
5. In many cases, setting the values of rxgain and txgain to 8.0 will achieve better echo elimination. Q: The analog card is not displayed by calling or incorrect.
A: the reason for this problem is that the zaptel driver version does not match the asterisk version. Asterisk and zaptel are usually released in pairs, and there is a ing between versions. Whether zaptel and Asterisk of different versions can be used together requires careful verification. Once a caller identification problem occurs, the simplest and most reliable verification method is to use the original trixbox2.0 disc to reinstall the system and then verify the caller display. Q: The SIP service can call but cannot hear the sound.
A: This is generally caused by Nat problems.
If the asterisk is behind the NAT, the asterisk configuration is as follows:
--------------------------------------------------------------
Sip. conf: ; Externip = 200.201.202.203
Externhost = dmlink.dvrdns.org
Localnet = 192.168.0.0/255.255.0.0 RTP. conf Rtpstart = 1, 10000
Rtpend = 20000 NAT Gateway port ing: SIP port 5060 and RTP Port 10000-20000 -------------------------------------------------------------- Externip or externhost needs to be configured. This is the public IP address of Asterisk and localnet needs to be configured. In this way, when Asterisk determines the external sip SESSION (localnet is used locally and the other party is not localnet ), the connection information is set to an external IP address in the SDP of the invite response (200 OK), and the media description port is allocated from [rtpstart, rtpend. In this way, after the SIP client receives the invite response (200 OK), it will know the RTP public IP + port of the asterisk. The voice stream of the SIP client is sent to this address, NAT Gateway (Port ing) forward to asterisk. How does asterisk determine the public RTP address of the SIP client after the SIP client is in a NAT state? After analyzing the asterisk code, the Asterisk SIP does not implement the ice ing from ice (Interactive connectivity establishment) to sip, that is, it does not process a new attribute ALT (Candidate IP address and port) defined by ice in the SDP media block. Instead, it adopts a simple method, that is, when Nat = Yes, from where to where (Sip also uses this policy ). Therefore, after Asterisk receives the RTP packet from the SIP client, asterisk records the source address and port of the packet (the Public IP address and port after the client-side Nat translation ), the voice stream of Asterisk will be sent to this address. The NAT gateway on the client side is automatically forwarded to the Intranet address of the corresponding asterisk. In this way, a SIP client (without ice) after Nat can already dial an asterisk after Nat configured with port ing. Q: Does asterisk support fax?
A: Yes. If you are installing trixbox, two fax sending and receiving commands rxfax and txfax are included in the system. If you install and compile your own asterisk, you need to install the rxfax and txfax apps on your own. Q: An error occurred while sending and receiving the fax.
Q: How to use a fax?
A: Dingming company website (http://www.dmlink.net/dowload2.asp has a pair of asterisk? .doc "explained in detail asterisk Fax sending and receiving problems and corresponding patches. Users who need it can download it. Q: Does asterisk Support No. 7 signaling?
A: asterisk itself does not support signaling 7. However, there are many open-source third-party modules. Dingming has released a strictly tested signaling module No. 7. Users who need it can download the trial from Dingming website.
Http://www.dmlink.net/dowload2.aspfile name: asterisk 7 signaling Module Q: Does asterisk support h323?
A: asterisk does not support h323. The asterisk-addon module includes several 323 modules. Dingming has launched a strictly tested h323 module. Users who need it can download the trial from Dingming website.
Http://www.dmlink.net/dowload2.aspfile name: asterisk ooh323c Module
Q: Can I insert multiple cards into one machine?
A: Yes.
Q: What kind of machine is required for the asterisk card?
A: Generally, it has a lot to do with the specific operating system, whether echo cancellation is enabled, whether encoding conversion is performed, and whether recording is performed.
A typical IVR application does not include recording. 4e1 must have at least P4 2.8 GB memory or above.
One SS7 to h323 application, 4e1 is recommended for dual Xeon 2.8g. Q: Is there a problem with the simulated card external line mounting detection?
A: Set in Zapata. conf:
Busydetect = Yes Q: g729 Protocol call Softswitch no return tone
A: g729 module issues. Check whether show translation g729 is correctly installed. If not, check whether/usr/lib/asterisk/modules/codec_g729.so exists. G729 is not self-contained and needs to be installed on its own.
If yes, use the g729 protocol to call the asterisk to enable the sound to be heard correctly. If you cannot hear it, the version is incorrect and cannot work properly. It should be normal if you hear it. Q: You can only hear the ring tones.
A: remove the t parameter from the dial (XXX,... t...) command. The t parameter forces asterisk to generate a return tone. Q: The number card pri call is directly hung up by the switch
A: There may be restrictions on relay numbers. You need to set callid to the specified number. You can use the setcallerid function in the dialing script.
Add zapatal. conf
Pridialplan = Local
Prilocaldialplan = Local Related Debugging commands:
PRI debug span 1 |
