[asterisk-users] Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean? Here is the output from 'sip show channels': Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 195.7.123.234 +180924402 3c3c4cee419 00102/0 alaw No Tx: ACK 9.9.94.9 6478517573 2752611-195 00101/1 ulaw No Rx: ACK 136.59.30.19 8787041796 76775e35788 00102/0 ulaw No Tx: ACK 9.9.95.13 9057047798 2752419-199 00101/1 ulaw No Rx: ACK 195.7.123.234 +011503733 25afde8070b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 71688696061 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 1700ab8b2ae 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011578435 0ecb33f75bb 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 71eac20715c 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 01b9eacf6de 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 744e7a3f501 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 0080443e6ad 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 6f3745a266d 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011221693 3b705a03141 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 4ab469132b7 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 0b2dcf2332b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 583bd73d09a 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011593222 4d237ba325e 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011639103 33f84238290 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011526778 72bd7b5f080 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011527693 0ffa93c642d 00102/2 unkn No (d) Rx: BYE gary___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with H option of Dial()
I also tried blind transfer with t option and it did not work. I added following into my dial plan contest: include => featuremap exten => 8111001001,1,Answer() exten => 8111001001,n,Wait(2) exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|tHL(12:61000:3)) exten => 8111001001,n,Hangup() It still does not work. I issue show features in CLI it show this: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# #1 Attended Transfer One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 What else do I need to do to make the features work? Gary Chen - Original Message - From: Gary Chen To: asterisk-users@lists.digium.com Sent: Tuesday, July 17, 2007 8:24 AM Subject: [asterisk-users] Problem with H option of Dial() I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H option in Dial() app. When press * during the call from caller side, Asterisk does not disconnect the call. The * just pass through. Here is my test dial plan: exten => 8111001001,1,Answer() exten => 8111001001,n,Wait(2) exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3)) exten => 8111001001,n,Hangup() It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss something? Or is it just a bug? Gary Chen -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with H option of Dial()
I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H option in Dial() app. When press * during the call from caller side, Asterisk does not disconnect the call. The * just pass through. Here is my test dial plan: exten => 8111001001,1,Answer() exten => 8111001001,n,Wait(2) exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3)) exten => 8111001001,n,Hangup() It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss something? Or is it just a bug? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Edit ulaw file
I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
- Original Message - From: "Darrick Hartman (lists)" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, June 24, 2007 11:25 AM Subject: Re: [asterisk-users] inband DTMF for g729 > Gang Chen wrote: >>> On 6/22/07, Gary Chen <[EMAIL PROTECTED]> wrote: >>>> We are using Level 3. At this point, changing carrier is not an option. >>>> >>> Gary, >>> >>> I use Level(3) with G729a and RFC2833. No problems, no requirement >>> for inband G729. >>> -- >>> Kristian Kielhofner >>> >> >> I can connect to Asterisk IVR using a SIP phone and send RFC2833 with >> g729. >> It works fine. But when test call from PSTN to Asterisk, if I set >> dtmf=auto >> with g729, I got warning saying something like * does not support inband >> for g729 and sutomaticlly switch to rfc2833. If I set dtmf=g729, there >> is >> no warning but I have the same problem. This tells me that Level3 does >> use >> inband for g729 or maybe I am doing something wrong . >> >> Gary > > Gary, > > I'll restate what Kristian just said above. You do NOT need inband for > Level 3. Set dtmf=RFC2833. > > Do you have the correct g729 codec licenses installed? This may be more > of a transcoding issue than anything else. > > Darrick > -- > Darrick Hartman > DJH Solutions, LLC > http://www.djhsolutions.com > We have not yet purchase the g729 codec licenses. I want to test it out first before we buy any license. I download g729 from Internet. I did set dtmfmode=rfc2833. It worked if I use an SIP phone connect to Asterisk using g729 and send dtmf tone using rfc2833. But not from PSTN through Level 3 . Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
We are using Level 3. At this point, changing carrier is not an option. - Original Message - From: "Matthew Fredrickson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, June 22, 2007 3:20 PM Subject: Re: [asterisk-users] inband DTMF for g729 > Sounds like you need a new SIP carrier. G.729 has a way of > destroying inband DTMF tones. > > --- > Matthew Fredrickson > Software Engineer > Digium, Inc. > > On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: > >> Does anybody know why Asterisk does not support inband DTMF for G.729? >> Our SIP carrier use inband dtmf for G.729. This causes problem for >> us to use it for our Asterisk IVR system. >> >> Any suggestion to solve this problem? >> >> Gary >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi with java?
- Original Message - From: "Matthew Pease" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, June 07, 2007 5:32 PM Subject: [asterisk-users] agi with java? Hi all - Searching for java agi in the mailing list archives turns up ancient posts. Anyone else using java for their AGI? How well is it working & what are you using? My script is pretty simple, and I could write it with perl easy enough, but I just would feel better if I can keep most programming code for our system in java. Thank you- Matt Our IVR system is written in Java. Our customer can call to check their account balance and pay their bill through the phone. It has been running on an old P3 machine for several years without single problem. The version of Asterisk is pretty old. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Track Agent login/logoff status
Asterisk 1.2.17 Right now I am using AGENTBYCALLERID_ variable to track agent login/logoff status. When an agent log into queue using AgentCallbackLogin(), AGENTBYCALLERID_ is set to agent phone number. When agent logoff, I use SetGlobalVar() to empty AGENTBYCALLERID_. This way by checking this variable, I can tell wether an agent is login or logoff in my dialplan. But sometime AGENTBYCALLERID_ is set to empty even if the agent is still in login mode. My guess is that somewhere asterisk reset this variable. Here are my questions: 1) Can I use this variable (AGENTBYCALLERID_) to track the status of agent for login and logff purpose? 2) What other event in Asterisk can also change value of this variable beside AgentCallbackLogin() function? 3) Any better way to do this? 4) Somebody metioned another function called AGENT. But I can not find it in my Asterisk. Does this function really exist? I know I can use AMI to check status of agent but I only need the agent status inside my dialplan. Gary ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
