[asterisk-users] High Availability - Shared Database - Ideas?
I am investigating High Availability solutions for my front end servers. I got into a discussion regarding "replicated local databases" versus " a single fiber connected shared database" on an EMC. Is anyone running a shared database on a SAN? Care to comment on your findings... Thanks, Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] convert from wav or mp3 to gsm
I use this all the time and am very pleased with the results... sox -r 8000 -c 1 resample -ql From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, March 30, 2010 3:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] convert from wav or mp3 to gsm AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files. You can use LAME and SOX to change files between these formats. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Tuesday, March 30, 2010 3:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] convert from wav or mp3 to gsm Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
I have used Rhinos for a while and they are very stable and work well with
asterisk... You need a T1 port port though
I also just bought a Xorcom and that is working very well too... (This is USB
so no need for a hardware card)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
Sent: Sunday, March 28, 2010 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
wrote:
> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>> Can anyone recommend a 24 fxs port voip gateway that has worked well
>> with asterisk? I have a couple of analog handsets that I want to
>> hookup to my asterisk server? Any tested and tried product recommendations
>> are welcome.
>> Thanks.
>
> Adtran channel banks are a great "trailing edge" technology. You can
> get them off Ebay for pennies on the original dollar and they are
> built like a tank.
>
> ("voip gateway" is not very specific. If you meant SIP or IAX, you
> might want to specify which.)
I've actually had decent success with the GXW-4024 (FXS <-> SIP) from
Grandstream which is probably one of the cheapest 24 FXS port boxes you'll find
out there.
-- James
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Re: [asterisk-users] Not hearing Telco Operator messages
chan_dahdi.conf: = ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2009-12-04 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no relaxdtmf=yes rxgain=3.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A108 port 1 [slot:4 bus:8 span:1] switchtype=national context=from-xo-nvfleetone3 group=2 echocancel=yes signalling=pri_cpe channel =>1-23 ;Sangoma A108 port 2 [slot:4 bus:8 span:2] context=from-xo-transp154760dl group=1 echocancel=yes signalling=em_w channel => 25-48 /etc/dahdi/system.conf = #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2009-12-04 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A108 port 1 [slot:4 bus:8 span:1] span=1,0,0,esf,b8zs bchan=1-23 echocanceller=mg2,1-23 hardhdlc=24 #Sangoma A108 port 2 [slot:4 bus:8 span:2] span=2,0,0,esf,b8zs e&m=25-48 echocanceller=mg2,25-48 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Friday, March 26, 2010 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not hearing Telco Operator messages Can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf. -- Zeeshan On 2010-03-26 9:06 AM, "Robert Grignon" wrote: I have 1 PRI and 1 E&M Wink Circuit. If I call a non working number and route it through the PRI, I get the following: "You have reached a non-working number." If I call a non working number and route it through the E&M Wink Circuit, I get the following: A core show channels shows the state as ringing... Is there something I have to do to get the "You have reached a non-working number." message to play back? Thanks, Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not hearing Telco Operator messages
I have 1 PRI and 1 E&M Wink Circuit. If I call a non working number and route it through the PRI, I get the following: "You have reached a non-working number." If I call a non working number and route it through the E&M Wink Circuit, I get the following: A core show channels shows the state as ringing... Is there something I have to do to get the "You have reached a non-working number." message to play back? Thanks, Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phone Assistance
Yes it does. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, March 17, 2010 8:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Door Phone Assistance Does a regular phone work on that port of the channel bank? On Wed, Mar 17, 2010 at 5:00 PM, Robert Grignon wrote: > I have two Viking E10 Door phones and a Rhino FXS channel bank... > > I have the channel set to immediate=yes and defined a custom context... > > When I press the button on the door phone, the inside phone rings and > I can hear the person talk through the door phone... The problem is I > cant hear anything through the speaker of the door phone... > > I know the speaker works because I do hear the initial ringing but > that's it... > > Could this be a voltage issue? I tried two different Viking Units... > > Thanks for any assistance. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Door Phone Assistance
I have two Viking E10 Door phones and a Rhino FXS channel bank... I have the channel set to immediate=yes and defined a custom context... When I press the button on the door phone, the inside phone rings and I can hear the person talk through the door phone... The problem is I cant hear anything through the speaker of the door phone... I know the speaker works because I do hear the initial ringing but that's it... Could this be a voltage issue? I tried two different Viking Units... Thanks for any assistance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xorcom Astribank Versus Rhino ChannelBank
I am very familiar Rhino Channel Banks and what to expect from them. I am intrigued by the Xorcom USB Channel Banks simply because I don't have to burn a hardware port... Can anyone comment on the Xorcom Astribank (24 FXS channels) and how well it works in an asterisk environment? I appreciate any info... Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling Segmentation Faults / Crashes
Just curious to know how most of you deal with segmentation faults / core dumps These are a few of the things I have seen in regards to people dealing with random crashes: 1. Apply Base OS updates 2. Recompile with DEBUG_THREADS and DON'T_OPTIMIZE turned on and look for the cause of the seg fault in GDB next time it happens 3. Upgrade to a late version of asterisk I have been running version 1.4.26 for a while and it has been relatively stable except for the random segfaults (once every few weeks). No errors or anything obvious in the logs, Server vitals are fine, etc... I am contemplating upgrading but I hate to knee jerk because sometimes you can introduce new problems with upgrading... My front-end servers are pretty busy throughout the day and usually have about 40 concurrent calls (80 channels) going at a time... Just looking for some "best practices" in regards to this. Thanks, Robert -- _ -- 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] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75
Come on, was that necessary? He was asking for help and considers it an important issue... If you want to chastise the guy at least offer up a solution for him... Sam - I have never tried this solution but Sangoma has a reference to this. http://wiki.sangoma.com/files/wanpipe-linux-asterisk-tutorials/How_to_Re duce_Asterisk_System_Loads.pdf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, February 08, 2010 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Do you want the advice in ALL CAPS? On 02/08/2010 11:42 PM, Muro, Sam wrote: > Hi Team > > Can someone advice me on how i can lower the load average on my > asterisk server? > > dahdi-linux-2.1.0.4 > dahdi-tools-2.1.0.2 > libpri-1.4.10.1 > asterisk-1.4.25.1 > > 2 X TE412P Digium cards on ISDN PRI > > Im using the system as an IVR without any transcoding or bridging > > ** > top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, > 62.55, > 55.75 > Tasks: 149 total, 1 running, 148 sleeping, 0 stopped, 0 zombie Cpu0 > : 10.3%us, 32.0%sy, 0.0%ni, 57.3%id, 0.0%wa, 0.0%hi, 0.3%si, > 0.0%st > Cpu1 : 10.6%us, 34.6%sy, 0.0%ni, 54.8%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu2 : 13.3%us, 36.5%sy, 0.0%ni, 49.8%id, 0.0%wa, 0.0%hi, 0.3%si, > 0.0%st > Cpu3 : 8.6%us, 39.5%sy, 0.0%ni, 51.8%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu4 : 7.3%us, 38.0%sy, 0.0%ni, 54.7%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu5 : 17.9%us, 37.5%sy, 0.0%ni, 44.5%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu6 : 13.3%us, 37.2%sy, 0.0%ni, 49.5%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu7 : 12.7%us, 37.3%sy, 0.0%ni, 50.0%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 3961100k total, 3837920k used, 123180k free, 108944k buffers > Swap: 779144k total, 56k used, 779088k free, 3602540k cached > >PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 683 > root 15 0 97968 36m 5616 S 307.7 0.9 41457:34 asterisk > 17176 root 15 0 2196 1052 800 R 0.7 0.0 0:00.32 top > 1 root 15 0 2064 592 512 S 0.0 0.0 0:13.96 init > 2 root RT -5 000 S 0.0 0.0 5:27.80 migration/0 3 > root 34 19 000 S 0.0 0.0 0:00.11 ksoftirqd/0 4 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 > root RT -5 000 S 0.0 0.0 1:07.67 migration/1 6 > root 34 19 000 S 0.0 0.0 0:00.09 ksoftirqd/1 7 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 > root RT -5 000 S 0.0 0.0 1:16.92 migration/2 9 > root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/2 > 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 > root RT -5 000 S 0.0 0.0 1:34.54 migration/3 12 > root 34 19 000 S 0.0 0.0 0:00.15 ksoftirqd/3 13 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 > root RT -5 000 S 0.0 0.0 0:54.66 migration/4 15 > root 34 19 000 S 0.0 0.0 0:00.01 ksoftirqd/4 16 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/4 17 > root RT -5 000 S 0.0 0.0 1:39.64 migration/5 18 > root 39 19 000 S 0.0 0.0 0:00.21 ksoftirqd/5 19 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/5 20 > root RT -5 000 S 0.0 0.0 1:06.27 migration/6 21 > root 34 19 000 S 0.0 0.0 0:00.03 ksoftirqd/6 22 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/6 23 > root RT -5 000 S 0.0 0.0 1:23.24 migration/7 24 > root 34 19 000 S 0.0 0.0 0:00.17 ksoftirqd/7 25 > root RT -5 000 S 0.0 0.0 0:00.00 watchdog/7 26 > root 10 -5 000 S 0.0 0.0 0:25.70 events/0 27 root > 10 -5 000 S 0.0 0.0 0:37.83 events/1 28 root > 10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10 > -5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 >000 S 0.0 0.0 0:16.45 events/4 > * > > Thanks > Sam > > > -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- 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] Can an agent Login to a queue and be paused
Not a bad idea... We use queuemetrics and the login is done via Web GUI.
I could easily just send it to pause upon login...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mariano
Lecuona
Sent: Monday, February 08, 2010 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can an agent Login to a queue and be
paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a
macro only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten => _211,1,Answer()
exten => _211,n,NoCDR()
exten =>
_211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})} > 1
]?4:5) ; Check that the physical extension is free
exten => _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)})
; Ask for agent password and log the agent on
exten => _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put
Agents into Initial Paused State on the Queue
exten => _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten => s,1,PauseQueueMember(|Agent/${ARG1})
exten => s,n,MacroExit
2010/2/8 Lenz Emilitri
I'm not sure if this works for newer versions of Asterisk, but
on old ones, you could pause an agent and THEN log him on, and he'd be
paused.
l.
2010/2/4 Robert Grignon
I thought there was an option for this but cant find
it
We have a busy callcenter and I would like the agents to
log in and be
in a paused state upon login... Right now they login and
they are
instantly receiving a call
Thanks for the input...
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] Can an agent Login to a queue and be paused
The agents tried that but it did not work... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Monday, February 08, 2010 4:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can an agent Login to a queue and be paused I'm not sure if this works for newer versions of Asterisk, but on old ones, you could pause an agent and THEN log him on, and he'd be paused. l. 2010/2/4 Robert Grignon I thought there was an option for this but cant find it We have a busy callcenter and I would like the agents to log in and be in a paused state upon login... Right now they login and they are instantly receiving a call Thanks for the input... -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Can an agent Login to a queue and be paused
I thought there was an option for this but cant find it We have a busy callcenter and I would like the agents to log in and be in a paused state upon login... Right now they login and they are instantly receiving a call Thanks for the input... -- _ -- 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] Analog Chanel locking up
I have asterisk 1.6.1.10 and a Rhino CB24 Channel Bank... A few channels seem to have locked up... If I plug an analog phone in the port, I get either dead air or a busy tone... Is there any way to reset this channel without restarting asterisk? ___ -- 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] For you sangoma users
I was dealing with an issue for a few weeks with my Gateway randomly crashing (Didn't matter what version of asterisk, sangoma firmware, etc)... I finally hooked up a modem cable to serial console and was able to catch the crash.. Wanpipe was causing it I spoke with Sangoma and they said dahdi-2.2 broke the RBS link... I was told to upgrade to 3.4.6.9 (not publicly published but on their ftp site) I have not had any crashes as of yet ___ -- 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] Polycom Phones
Sorry if this is off topic I have a "loud talker" in our call center and was asked if I can make his voice louder to make him talk softer :-) Does anyone know if you can do that with Polycom 430's I found voice.gain.tx.headset but wasn't sure if that will make his voice louder to the calling party or to himself... ___ -- 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] asterisk 1.4.26.3 makes kernel panic
What Hardware are you using? What OS are you running? If your getting a kernel panic you can install a crashkernel (kdump) and upon receiving a kernel panic it will reboot to a crashkernel, capture the crashinfo and safely reboot the system. You can then use the "crash" utility to analyse the information. However, first thing I would try is to change the slot that your hardware card is plugged into... The kernel is complaining about an inturrupt issue and its possible its conflicting with something else... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, November 18, 2009 4:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk 1.4.26.3 makes kernel panic Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 0130 ebx: ecx: 00220028 edx: 0978 esi: 346e5802 edi: ebp: c3b45500 esp: caf05bfc ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process asterisk (pid: 7631, ti: caf04000 task: f7987000 task.ti: caf04000) Stack: ... Call Trace: ... Code: ... EIP: [] ss:esp 0068:caf05bfc Kernel panic - not syncing: Fatal exception in interrupt As you can see, kernel is 2.6.23 but it's the same as on another server I have but with asterisk 1.2. Kernel 2.6.23 does not panic with asterisk 1.2. The hardware on both servers is identical (isdn cards, pri card, etc.) except for the motherboard. How can I further diagnose the problem (besides, I think I can easily reproduce it as soon as I set it up as the "production" server)? What can be causing the kernel panic? BTW, the last "log entry" I have in /var/log/asterisk/full is: [Nov 18 10:08:20] VERBOSE[6627] logger.c: -- dialparties.agi: Checking CW and CFB status for extension 6169 (this system runs freepbx) Help greatly appreciated. Vieri ___ -- 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
Re: [asterisk-users] my kernel is dazed and confused
This can also be caused by IRQ conflicts. You could try a different slot to see if it clears it up -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, November 12, 2009 1:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] my kernel is dazed and confused On Thu, Nov 12, 2009 at 09:31:11AM -0500, Dr. Michael J. Chudobiak wrote: > Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown > reason a0 on CPU 0. > Nov 12 08:54:27 steerpike kernel: You have some hardware problem, > likely on the PCI bus. > Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to > continue NMI - Non Maskable Interrupt. This is a rather generic error message. Search a bit to see how to make some more sense of the messages following it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Crashing need some ideas
I am about out of ideas I am not able to keep this gateway stable. I am crashing about 2 times a day Is there a way to capture the crash data? I have kdump configured on the server but it seems to be a hard lockup and not a kernel panic I have tried the following: Asterisk 1.4.26.2 Asterisk 1.6.1.8 Asterisk 1.6.1.9 Sangoma A108d firmware 35,38,39 We have a PRI and a T1 (em_wink)... I have called the circuit provider and verified settings with them as well... Thanks for any insight ___ -- 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] Anyone seen this before
Testing a new gateway and have a Rhino Channel Bank... Sending a test fax and everything works fine (Receive the fax fine) But I notice this in the log Google search didn't return much of anything... DAHDI hook failed returned -1 (trying 1): Device or resource busy ___ -- 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] Delayed answer when calling out
Tried it but didn't seem to change anything... You were meaning something like Dial(DAHDI/g2/w1611212,30,tTr) right ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 29, 2009 2:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Delayed answer when calling out Might or might not be relevant. Try dial(DAHDI/X/w#) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Thursday, October 29, 2009 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delayed answer when calling out I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear "thank you for calling. Please press a number" When placing a call through the LDT1 to the same number. I hear "...Please press a number" It is cutting off the "Thank you for calling" I also notice that I dont hear a ringback... I've tried the following: added "r" to the dial command (this does give me a ringback but its still cutting off the first few seconds) set "usecallerid=no" to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] Delayed answer when calling out
I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear "thank you for calling. Please press a number" When placing a call through the LDT1 to the same number. I hear "...Please press a number" It is cutting off the "Thank you for calling" I also notice that I dont hear a ringback... I've tried the following: added "r" to the dial command (this does give me a ringback but its still cutting off the first few seconds) set "usecallerid=no" to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] GUI for hunt groups?
www.voiceroute.org also has an open source unified communications manager (they also have a commercial version)... Very little support from the developers but I have deployed it in a few large call centers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, October 28, 2009 7:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] GUI for hunt groups? Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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
Re: [asterisk-users] Having a heck of a time
Upon further research I kind of answered my own question.. But I will share... If you are seeing multiple H.100 errors in your system log then the hardware echo canceler does not have a good clock source. On our more recent drivers 3.3.12 and up the first port that starts up will be the clocking source. So if your wanpipe1 is not connected then please configure your card to only start the first port that connects. If you have an older driver then the timing source is the first physical port on the card. So if you are not using the first physical port then please follow the steps below to set another port as a timing source. Please note that only one port can act as timing source for HWEC in a particular AFT102/104/108 card, in other words you can only set HWEC_CLCKSRC = YES for only one port for a card! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Wednesday, October 28, 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Yes I did that... I even recompiled dahdi-linux and tools after wanpipe install... Once I did that it recognized the card and said I could run "dahdi_genconf modules" which in turn would only load the cards that it seeing. I had the PRI running in slot 6. Once I unplugged the PRI I was able to get a response from dahdi_test I then wondered if it was looking for a circuit on channel 1 (this didnt make much sense because the PRI is getting timing from the telco and the port location should not matter) I then moved the PRI to channel 1 and dahdi_test returned the following: [r...@lin-vgw1 asterisk]# dahdi_test -vc 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8190.808 system clock sample intervals (99.985%) 8192 samples in 8190.288 system clock sample intervals (99.979%) 8192 samples in 8190.776 system clock sample intervals (99.985%) 8192 samples in 8190.872 system clock sample intervals (99.986%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8190.833 system clock sample intervals (99.986%) 8192 samples in 8190.960 system clock sample intervals (99.987%) 8192 samples in 8190.864 system clock sample intervals (99.986%) 8192 samples in 8190.744 system clock sample intervals (99.985%) 8192 samples in 8190.800 system clock sample intervals (99.985%) --- Results after 10 passes --- Best: 99.987 -- Worst: 99.979 -- Average: 99.984940, Difference: 99.984940 GO figure... I did notice this in the logs and am not sure what to make of the "CT_C8_A clock behavior does not conform to the H.100 spec" reference: Oct 28 13:26:19 lin-vgw1 kernel: wanpipe6: Lost of Signal is detected! (Unplugged circuit and moved to channel 1) Oct 28 13:26:20 lin-vgw1 kernel: wanpipe2: Lost of Signal is detected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:LOF : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:RED : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: T1 disconnected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: Enable transmit RAI alarm Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: AFT communications disabled! (Dev Cnt: 6 Cause: Link Down) Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: TDM Free Run Timing Enabled 1 ms Oct 28 13:26:23 lin-vgw1 kernel: wanpipe6:LOS : ON Oct 28 13:26:24 lin-vgw1 kernel: wanpipe2:LOS : ON Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The H100 slave has lost its framing on the bus! Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The CT_C8_A clock behavior does not conform to the H.100 spec! Oct 28 13:27:06 lin-vgw1 kernel: wanpipe5: Lost of Signal is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1:RAI : ON Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Open Circuit is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Lost of Signal is cleared! Oct 28 13:27:17 lin-vgw1 kernel: wanpipe1:RAI : OFF Oct 28 13:27:17 lin-vgw1 kernel: wanpipe5:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOF : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:RED : OFF Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: T1 connected! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT communications enabled! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT Global TDM Intr Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: Global TDM Ring Resync TDM = 0x1 Oct 28 13:27:21 lin-vgw1 kernel: ADDRCONF(NETDEV_CHANGE): w1g1: link becomes ready From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Wednesday, October 28, 2009 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Did you use ./Setup dahdi when installing the wanpipe drivers? http://wiki.sangoma.com/wanpipe-
Re: [asterisk-users] Having a heck of a time
Yes I did that... I even recompiled dahdi-linux and tools after wanpipe install... Once I did that it recognized the card and said I could run "dahdi_genconf modules" which in turn would only load the cards that it seeing. I had the PRI running in slot 6. Once I unplugged the PRI I was able to get a response from dahdi_test I then wondered if it was looking for a circuit on channel 1 (this didnt make much sense because the PRI is getting timing from the telco and the port location should not matter) I then moved the PRI to channel 1 and dahdi_test returned the following: [r...@lin-vgw1 asterisk]# dahdi_test -vc 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8190.808 system clock sample intervals (99.985%) 8192 samples in 8190.288 system clock sample intervals (99.979%) 8192 samples in 8190.776 system clock sample intervals (99.985%) 8192 samples in 8190.872 system clock sample intervals (99.986%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8190.833 system clock sample intervals (99.986%) 8192 samples in 8190.960 system clock sample intervals (99.987%) 8192 samples in 8190.864 system clock sample intervals (99.986%) 8192 samples in 8190.744 system clock sample intervals (99.985%) 8192 samples in 8190.800 system clock sample intervals (99.985%) --- Results after 10 passes --- Best: 99.987 -- Worst: 99.979 -- Average: 99.984940, Difference: 99.984940 GO figure... I did notice this in the logs and am not sure what to make of the "CT_C8_A clock behavior does not conform to the H.100 spec" reference: Oct 28 13:26:19 lin-vgw1 kernel: wanpipe6: Lost of Signal is detected! (Unplugged circuit and moved to channel 1) Oct 28 13:26:20 lin-vgw1 kernel: wanpipe2: Lost of Signal is detected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:LOF : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:RED : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: T1 disconnected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: Enable transmit RAI alarm Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: AFT communications disabled! (Dev Cnt: 6 Cause: Link Down) Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: TDM Free Run Timing Enabled 1 ms Oct 28 13:26:23 lin-vgw1 kernel: wanpipe6:LOS : ON Oct 28 13:26:24 lin-vgw1 kernel: wanpipe2:LOS : ON Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The H100 slave has lost its framing on the bus! Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The CT_C8_A clock behavior does not conform to the H.100 spec! Oct 28 13:27:06 lin-vgw1 kernel: wanpipe5: Lost of Signal is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1:RAI : ON Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Open Circuit is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Lost of Signal is cleared! Oct 28 13:27:17 lin-vgw1 kernel: wanpipe1:RAI : OFF Oct 28 13:27:17 lin-vgw1 kernel: wanpipe5:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOF : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:RED : OFF Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: T1 connected! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT communications enabled! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT Global TDM Intr Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: Global TDM Ring Resync TDM = 0x1 Oct 28 13:27:21 lin-vgw1 kernel: ADDRCONF(NETDEV_CHANGE): w1g1: link becomes ready From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Wednesday, October 28, 2009 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Did you use ./Setup dahdi when installing the wanpipe drivers? http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi ___ -- 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] Having a heck of a time
That was a good thought. I have 3 other gateways in production and I ran dahdi_test and zttest (older gateways) and they all said they were opening a "psedu" device -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BJ Weschke Sent: Wednesday, October 28, 2009 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon wrote: > This has been a rollercoaster ride > > Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) > > Where I stand right now, I have a PRI on the gateway and circuit is > working I can make calls through the gateway > > > > Here is my problem: > DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work > > [r...@lin-vgw1 asterisk]# dahdi_test > Opened pseudo dahdi interface, measuring accuracy... > > --- Results after 0 passes --- > Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: > 100.00 > > Also if I establish a call and run dahdi_monitor it doesnt look quite > like it is supposed to: > > [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv > > Visual Audio Levels. > > Use chan_dahdi.conf file to adjust the gains if needed. > > ( # = Audio Level * = Max Audio Hit ) > <(RX)> > <(TX)> > > I'm looking for some ideas here? > I could be way off, but I think the fact that it says it's opening the psuedo interface implies that it doesn't see your Sangoma card. You might want to try and check with Sangoma support to see what they have to say about that. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- 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] Having a heck of a time
This has been a rollercoaster ride Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) Where I stand right now, I have a PRI on the gateway and circuit is working I can make calls through the gateway Here is my problem: DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work [r...@lin-vgw1 asterisk]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... --- Results after 0 passes --- Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: 100.00 Also if I establish a call and run dahdi_monitor it doesnt look quite like it is supposed to: [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) <(RX)> <(TX)> I'm looking for some ideas here? Thanks ___ -- 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] hangup from which side
We have queuemetrics and it does that
Here is some of the logic - (Obviously this wont work for you right out
of the box but you should be able to decipher the logic...)
[qm-queuedial]
; We use a global variable to pass values back from the answer-detect
macro.
; STATUS = U unanswered
;= A answered(plus CAUSECOMPLETE=C when callee hung up)
; The 'g' dial parameter must be used in order to track callee
disconnecting.
; Note that we'll be using the 'h' hook in any case to do the logging
when channels go down.
; We set the CDR(accountcode) for live monitoring by QM.
;
exten => s,1,NoOp,Outbound call -> A:${QDIALER_AGENT}
N:${QDIALER_NUMBER} Q:${QDIALER_QUEUE} Ch:${QDIALER_CHANNEL}
exten => s,n,Set(CDR(accountcode)=QDIALAGI)
exten => s,n,Set(ST=${EPOCH})
exten => s,n,Set(GM=QDV-${QDIALER_AGENT})
exten => s,n,Set(GLOBAL(${GM})=U)
exten => s,n,Set(GLOBAL(${GM}ans)=0)
exten =>
s,n,Macro(queuelog,${ST},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT},C
ALLOUTBOUND,-,${QDIALER_NUMBER})
exten =>
s,n,Dial(${QDIALER_CHANNEL},300,gM(queuedial-answer^${UNIQUEID}^${GM}^${
QDIALER_QUEUE}^${QDIALER_AGENT}^${ST}))
exten => s,n,Set(CAUSECOMPLETE=${IF($["${DIALSTATUS}" = "ANSWER"]?C)})
; Trapping call termination here
exten => h,1,NoOp( "Call exiting: status ${GLOBAL(${GM})} answered at:
${GLOBAL(${GM}ans)} DS: ${DIALSTATUS}" )
exten => h,n,Goto(case-${GLOBAL(${GM})})
exten => h,n,Hangup()
; Call unanswered
exten => h,n(case-U),Set(WT=$[${EPOCH} - ${ST}])
exten =>
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},ABANDON,1,1,${WT})
exten => h,n,Hangup()
; call answered: agent/callee hung
exten => h,n(case-A)i,Set(COMPLETE=${IF($["${CAUSECOMPLETE}" =
"C"]?COMPLETECALLER:COMPLETEAGENT)})
exten => h,n,Set(WT=$[${GLOBAL(${GM}ans)} - ${ST}])
exten => h,n,Set(CT=$[${EPOCH} - ${GLOBAL(${GM}ans)}])
exten =>
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},${COMPLETE},${WT},${CT})
exten => h,n,Hangup()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus
Darilion
Sent: Friday, October 23, 2009 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb:
> When Asterisk establish a call through an outbound trunk, Is there any
> way I can know who hang up the call first? The caller or the party
called?
you could use the 'g' option of the Dial command together with some
logic in the hangup extensions
regards
klaus
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Re: [asterisk-users] Incorrect voice mail format on transfer
I also noticed that there was a version of asterisk that had a voicemail bug dealing with this... I am run 1.4.26.2 now and what was happening was if IMAP forwarded (and corrupted) a voicemail, the user would try to retrieve the message and the system would hangup on them.. The updated code seemed to acknowledged the corrupted message (if it happened again) and ignored it and went to the next message... You probably have a problem with people deleting temporary "out of office" messages as well... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 22, 2009 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect voice mail format on transfer Phew! So it's not just me! That's exactly the problem - not leaving the message but forwarding it (I suppose the correct term rather than transfer). Thanks - John On Thu, 2009-10-22 at 10:29 -0500, Robert Grignon wrote: > I did run into some issues with this as well. I ended up setting > format=wav and left it at that... It wasn't so much a problem with > someone leaving a message rather when someone was forwarding messages. > I would have used wav49 but people were having problems getting wav49 > to open on their PDA's > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. > Sullivan III > Sent: Wednesday, October 21, 2009 4:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Incorrect voice mail format on transfer > > I'm sorry - by the lab I meant the end points - it is the same server. > > I was not aware that IMAP only stored one format. If I change the > setting in voicemail.conf, do I still have to worry about the grievous > warning message about being sure to delete all messages not using that > format? I would think not but it's a dire enough message that I > thought I had better ask - John > > On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: > > It should be reproducible in some way, how was asterisk installed on > > the server its having a problem? If its from source compare the > > apps/app_voicemail.c from whats in production with whats getting > > compiled in the lab. > > > > > > when imap is used only one format is stored you could specify just > > one > > > format: > > format=wav49 > > > > On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III > > wrote: > > Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a > > multi-tenant environment with IMAP voice mail storage on > > Zimbra. One of > > our clients is having a problem when transferring voice mails > > from one > > mailbox to another (option 8 in the standard voice application > > menu) > > using their Snom 320 and 360 phones. > > > > The end results is the final recipient cannot listen to the > > voicemail. > > We also email the voicemails in this case (this client is not > > using the > > Zimbra email system yet) and they receive an attachment with a > > name such > > as "msg.wav49_gsm_wav". > > > > As strange as it sounds, it almost appears like Asterisk is > > trying to > > create a file with an extension of wav49|gsm|wav which is > > confusing not > > only the email attachment but also sox which cannot find such > > a format > > based upon file extension. Here is what I see > > in /var/log/asterisk/messages. > > > > First, the user doing the transfer: > > [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP > > Warning: SECURITY PROBLEM: insecure server advertised > > AUTH=PLAIN > > [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP > > Warning: SECURITY PROBLEM: insecure server advertised > > AUTH=PLAIN > > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed > > to > > reencode > /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error > occurred during file processing (have you installed support for all > sox file formats?) > > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail > > attachment will have no volume gain. > > [Oct 21 12:28:44] W
Re: [asterisk-users] Incorrect voice mail format on transfer
I did run into some issues with this as well. I ended up setting format=wav and left it at that... It wasn't so much a problem with someone leaving a message rather when someone was forwarding messages. I would have used wav49 but people were having problems getting wav49 to open on their PDA's -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Wednesday, October 21, 2009 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect voice mail format on transfer I'm sorry - by the lab I meant the end points - it is the same server. I was not aware that IMAP only stored one format. If I change the setting in voicemail.conf, do I still have to worry about the grievous warning message about being sure to delete all messages not using that format? I would think not but it's a dire enough message that I thought I had better ask - John On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: > It should be reproducible in some way, how was asterisk installed on > the server its having a problem? If its from source compare the > apps/app_voicemail.c from whats in production with whats getting > compiled in the lab. > > > when imap is used only one format is stored you could specify just one > format: > format=wav49 > > On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III > wrote: > Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a > multi-tenant environment with IMAP voice mail storage on > Zimbra. One of > our clients is having a problem when transferring voice mails > from one > mailbox to another (option 8 in the standard voice application > menu) > using their Snom 320 and 360 phones. > > The end results is the final recipient cannot listen to the > voicemail. > We also email the voicemails in this case (this client is not > using the > Zimbra email system yet) and they receive an attachment with a > name such > as "msg.wav49_gsm_wav". > > As strange as it sounds, it almost appears like Asterisk is > trying to > create a file with an extension of wav49|gsm|wav which is > confusing not > only the email attachment but also sox which cannot find such > a format > based upon file extension. Here is what I see > in /var/log/asterisk/messages. > > First, the user doing the transfer: > [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP > Warning: SECURITY PROBLEM: insecure server advertised > AUTH=PLAIN > [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP > Warning: SECURITY PROBLEM: insecure server advertised > AUTH=PLAIN > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed > to > reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail > attachment will have no volume gain. > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to > open > file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: > No such file or directory > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed > to > reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail > attachment will have no volume gain. > [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to > open > file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory > [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed > to > reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) > [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail > attachment will have no volume gain. > [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to > open > file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: > No such file or directory > [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed > to > reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) > [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail > attachment will have
[asterisk-users] Intermittent Low volume
Just looking for some ideas here... Single office with 1.4.26.2 - Frontend & 1.4.26.2 w/sangoma A108 Gateway I have been getting a few complaints about "caller cant hear me" or "I cant hear the caller" I've listened to the recordings and can verify what they are complaining about, with this being said, most calls are fine. I know there are alot of issues that can happen once the call leaves the office that I will never be able to address (vonage, cell phones, etc) but I am trying to see if there is anything I could do to help alleviate the issue on my end. I never really messed with rxgain and txgain and was starting to play with dahdi_monitor to see my gain levels... Do you all think this could be a gain level issue? Thanks for any input ___ -- 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] High Volume Call Center SIP versus IAX2
I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to make sure I am approaching the setup correctly. Thanks ___ -- 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
