search for: thowint

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

Hi, I have an queue stored in relatime and defined members called through LOCAL/.... I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. best regards Thomas

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

...erisk only checks the IP the call is coming from and uses the context you defined there. If you use type=user you will need to specify a username and a secret. -- Greetings.. V?ctor Toofic ------------------------------ Message: 23 Date: Tue, 23 Oct 2007 00:11:39 +0200 From: Thomas Winter <thowinter at googlemail.com> Subject: [asterisk-users] bristuff: music on hold but no dialoptions tT defined. To: asterisk-users at lists.digium.com Message-ID: <200710230011.39639.thowinter at googlemail.com> Content-Type: text/plain; charset="utf-8" Hi, Iam dialing from NT ptp to S...

Hi, the documentation of sip.conf is telling me this: ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas

Hi, I have to set up an Asterisk with an 4-port BRI card. Hylafax should send and receive fax. Will this work reliable? Any recommandations for an 4-port BRI card? Other alternatives except analog fax units? thanks for your help best regards Thomas

Hi, If Iam doing UPDATE SQL statements I got an overload for connection. am doing everytime an Disconnect ${connid}) but this is ignored. any idea? best regards Thomas

Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? best regards Thomas

If i do an asterisk -rx "moh reload" MoH stops and restarts on exsisting channels. If I do an moh reload through the Manager Interface Sound is dead on exsisting channels. Any other idea for an workaround? Hi, I have problems with 1.2.14 and musiconhold.conf and realtime. I have to do moh reload at CLI to use the classes stored in mysql. Otherwise nothing is found if using

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983]

Hi, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best regards Thomas

Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas

Hi list, how can I set up an peer, so that behind one IP (NAT) multiple devices can access to this single peer to make outbound calls. Some of these multiple devices will be SIP phones and these SIP phones are trying to make registrations to this peer. best regards Thomas

Hi, when I audio studio should produce an sound file to play back with Asterisk. Whats the best format they should deliver the audio file? Sample Size : 16-bit (2 bytes) Sample Encoding: signed (2's complement) Channels : 1 Sample Rate : 8000 thanks Thomas

Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks

Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas