similar to: Failed to authenticate on INVITE

Displaying 20 results from an estimated 900 matches similar to: "Failed to authenticate on INVITE"

2008 Jan 16
3
volume problem
Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango
2009 Feb 24
7
multiple asterisks in a server
Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango
2008 Feb 20
3
[LLVMdev] Bug? Coalescing & Updating Subreg Intervals
I have a question about what is going on at line 754 of SimpleRegisterCoalescing.cpp. The comment says we are updating the live intervals for subregisters. This happens when we coalesce to a physical register. Now, I read that as, "merge in the range information from the eliminated live interval to the subregister live interval," but that appears to not be what happens. In my case,
2008 May 05
3
simple realtime question
HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango
2007 Feb 22
2
fax support
Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to
2009 May 29
2
regarding to field of accountcode
Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango
2012 Aug 06
1
[Announce] Samba 3.6.7 Available for Download
=================================================================== "Change is such hard work." Billy Crystal =================================================================== Release Announcements ===================== This is is the latest stable release of Samba 3.6. Major enhancements in Samba 3.6.7 include: o Fix resolving our own "Domain Local" groups
2012 Aug 06
1
[Announce] Samba 3.6.7 Available for Download
=================================================================== "Change is such hard work." Billy Crystal =================================================================== Release Announcements ===================== This is is the latest stable release of Samba 3.6. Major enhancements in Samba 3.6.7 include: o Fix resolving our own "Domain Local" groups
2007 Sep 22
1
prepaid application recommendation
Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango
2008 Mar 13
1
asterisk out of service
Hi all, I got the following message in the log yesterday. After that, no more in/out bound call can be made. What is the meaning of the message? ango [Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2367-d8062fb0! [Mar 12 09:26:17] ERROR[29565] chan_sip.c: SIP transaction failed: 5999e928603c878945d3e7811d2393e8 at 210.14.27.50 [Mar 12 09:33:15] ERROR[29565]
2009 Jan 15
1
call transfer in CDR
Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango
2009 Mar 28
2
hum noise
HI, We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango
2009 Apr 27
1
music on hold using mms
Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango
2009 May 21
1
interruption in queue
HI, I want to allow user to press 0 to the voicemail if the user don't want to wait in the queue. Below is what I set but it doesn't work. Anyone can help? ango file: features.conf [applicationmap] opervm => 0,self/both,Macro,opervm file: extensions.conf ... exten => 5555,n(queue),Set(DYNAMIC_FEATURES=opervm) exten => 5555,n,Queue(5555|tThH|||180) ... [macro-opervm] exten
2009 Oct 22
1
queues autopause
Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in
2008 Feb 13
6
restart asterisk daily
Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango
2006 Nov 22
0
help in Call parking......
Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial
2013 Dec 17
2
Setting up a lustre zfs dual mgs/mdt over tcp - help requested
Hi all, Here is the situation: I have 2 nodes MDS1 , MDS2 (10.0.0.22 , 10.0.0.23) I wish to use as failover MGS, active/active MDT with zfs. I have a jbod shelf with 12 disks, seen by both nodes as das (the shelf has 2 sas ports, connected to a sas hba on each node), and I am using lustre 2.4 on centos 6.4 x64 I have created 3 zfs pools: 1. mgs: # zpool
2008 Nov 17
3
Gigabit Lan doesn't work
Hi all, I have installed Centos completely. However, the LAN doesn't work. Below is the message after I issue. How can I make it work? 00:19.0 Ethernet controller: Intel Corporation 82567V-2 Gigabit Network Connection Thanks!
2008 Oct 29
1
SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown"
Please help with this strange issue. When "sip show peers" returns status "Unknown" the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username Host