similar to: Using Sangoma Call Progress Analysis behind NAT router

Displaying 20 results from an estimated 2000 matches similar to: "Using Sangoma Call Progress Analysis behind NAT router"

2009 Apr 17
1
Sangoma A104d and Adtran 850 problems
I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one
2009 Jul 21
1
Dialplan step that I do not have
I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom
2010 May 10
1
More clarification on outbound sip channels.
Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus
2011 Apr 14
1
Microsoft Lync server and Asterisk access
We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to
2008 Nov 03
0
busylevel question
I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it
2008 Nov 06
0
Asking again about busylevel
I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then
2008 Nov 20
1
Playback using AMI
Is there a way to inject sound from a sound file into an established call using AMI? I have an established call from which I can record either or both legs. I can additionally "spy" on the call. Is there any way I can play a sound file into the call and not loose the ability for the people to continue talking while listening to the sound file? -- Jim Dickenson mailto:dickenson at
2008 Oct 14
1
Help With AMI
I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser
2009 Jun 24
3
dahdi-linux-2.2.0 compile problem
I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this?
2011 Jan 06
2
Benefit of PRI vs SIP trunk calls
Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can
2009 Jan 28
1
Scope of variable
I have this extension: exten => 1322,1,Answer() exten => 1322,n,Set(CfMC_AMDValue="NotChecked") exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD) exten => 1322,n,AMD() exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS}) exten => 1322,n(NOAMD),Wait(1) exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} & ${CfMC_AgentToUse}
2009 Feb 11
0
ChanSpy problem
I have an extension defined like this: exten => do_monitor,1,Answer() exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}') exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX) exten => do_monitor,n,Hangup() I use an AMI packet like this: Action: Originate Channel: Agent/1001 Exten: do_monitor Context: cfmc_cdi_private Priority: 1 Variable: CfMC_ActionID=callE1334 Variable:
2009 Feb 04
0
Stopping chanspy
I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same
2009 Jun 17
2
What causes this error?
[2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295] == Primary D-Channel on span 1 up [2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a UA, but i'm in state 7 I noticed the above error many days after this at around 2AM. This
2009 Jun 26
0
Problem with RetryDial
I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and
2009 Mar 02
3
How to set PRI line timeout value
I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then I go to extension h and have
2010 Mar 07
1
Caller Presentation Confusion
I have been fighting with the ability to set the caller ID when I make outbound calls via a PRI line as well as via my SIP provider. The more I play around the less I understand. There is a setting in chan_dahdi.conf that seems to say do not pay attention to the CALLERPRES value and just allow the ID to be set. This setting is usecallingpres. If this is set to yes then the value of CALLERPRES
2009 Feb 04
1
Stopping chanspy followup
I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches
2008 Dec 05
2
AMI interface problem
I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:
2009 Aug 18
0
Date/time in queue_log
1250628579|MANAGER|S1140|Agent/265|ADDMEMBER| Above is an example of a line from the file. Is there an option to have this date/time output in a way that can be read without conversion? The date/time in full and messages is formated so it can be read directly. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/