No audio on call forward after upgrade from Asterisk 1.4 to 1.6

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I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single issue that I can't explain. I have an extension that if you call it, it will play a sound file and hangup. Pretty simple stuff. Below is the extensions.conf entry for this extension. exten => 849,1,Playback(custom/ceh-meetingmsg) exten => 849,n,Hangup

Asterisk Users 5 years ago 5 Answers

Asterisk and Cisco Unified IP Phone 9971

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I am having a weird issue with a Cisco 9971 phone. I managed to get most of it working, including the side car, however one of the issues is that there seems to be some sort of side tone / beep occurring roughly every 13 seconds or so, as if the phone is activated with call waiting. However none of this is activated and is still making the annoying beeping side tone. The phone does require that one runs with tcp=enable and transport=tcp, thus turning on the presence information, which from the logs seems to be refreshing roughly every 10 - 15 seconds. However the "Patch" has…

Asterisk Users 5 years ago 1 Answer

Asterisk Voicemail Prompts Fuzzy And Quiet

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Strange issue that I can't figure out and I am hoping someone may have some ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it runs like a top and I am not going to mess with it). *B rsyncs config from *A. *A dies. I bring up *B and it all works fine, except for one issue. Calls to voicemail are garbled and low. Phone to phone and phone to gateway work perfect. If I get voicemail emailed to me, it sounds perfect. If I call into voicemail, the prompts and the message are garbled. There is no packet loss or any…

Asterisk Users 5 years ago 0 Answers

Wifi + SIP + Asterisk

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In my old office, for conference purpose , gotomeeting was used. also for the lecture delivery, same gobomeeting was used, most the time , we need to listen voice only. also, we use to share desktop screen. But as far as I know SIP is the standard for video telephony. SIP can handle video +Audio. now, I am thinking that, I can give solution like selling a server which has Asterisk over it. AFAIK, Asterisk can handle VoIP calls. Now system will load some GUI application from there he can add remove users. Now at the same time I want to give small device which has Wifi + LCD…

Asterisk Users 5 years ago 0 Answers

help with dialplan

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Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk is 1.6.2.11. Below is the extensions.conf file. [globals] QPHONE0=SIP/10 QPHONE1=SIP/11 QPHONE2=SIP/12 QPHONE3=SIP/13 QPHONE4=SIP/14 QPHONE5=SIP/15 QPHONE6=SIP/16 QPHONE7=SIP/17 ACAPHONE0=SIP/20 ACAPHONE1=SIP/21 ACAPHONE2=SIP/22 ACAPHONE3=SIP/23 ACAPHONE4=SIP/24 ACAPHONE5=SIP/25 ACAPHONE6=SIP/26 ACAPHONE7=SIP/27 GMNETPHONE0=SIP/30 GMNETPHONE1=SIP/31 GMNETPHONE2=SIP/32 GMNETPHONE3=SIP/33 GMNETPHONE4=SIP/34 GMNETPHONE5=SIP/35 GMNETPHONE6=SIP/36 GMNETPHONE7=SIP/37 EXTERNPHONE0=SIP/150 CPHONE1=SIP/16780000000 CPHONE2=SIP/17700000000 EMERGENCY=0 EMERGENCY_TRUNK=DAHDI/G1 ; Change this for production use: EMERGENCY_NUM=6789542133 [from-pstn] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming1,s,1) [from-pstn1] exten => s,1,Set(FROM_DID="6780000000)…

Asterisk Users 5 years ago 22 Answers

Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

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I've recently had a fairly prolonged SIP registration attack, 18 hours in this case and often with 200 attempts per second, and suspect I've had a number of these in the past. The main symptom I noticed previously was, because Asterisk was responding to each registration request it received, it was very quickly using up my 448 kbps upload limit for my home ADSL connection: any further traffic (i.e. anything I did) was then experiencing significant packet loss. Anyway, I've now implemented the "7 steps to better Asterisk security" that I found on the Digium website (deny/permit, alwaysauthreject etc.), and have been looking at fail2ban. However, when I…

Asterisk Users 5 years ago 10 Answers

How to Billing for MeetMe Conference In Asterisk

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MeetMe has cdr for each attendant. but the fee is always paid by the moderator. not by each one. and the the members in the conference are dynamic changed. In this scenario, how to billing for MeetMe conference? and i want to hungup all the calls when the account fee of the moderator is not enough. Thanks for your help!  

Asterisk Users 5 years ago 0 Answers

Asterisk Routing To SoftSwitch

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I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik

Asterisk Users 5 years ago 5 Answers

Play Files To Caller In Asterisk

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I want to be able to allow a caller to dial a DDI system to verify identity etc (this is all done), I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the file, and then jump back to the original Asterisk dialplan entry to start again. However, If I want to redirect, then this external event would need to know their…

Asterisk Users 5 years ago 10 Answers