[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

[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 bei

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 --

Re: [asterisk-users] Incorrect voice mail format on transfer

ly 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 was

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 ;

[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 DA

Re: [asterisk-users] Having a heck of a time

um.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 > > Bui

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

Re: [asterisk-users] Having a heck of a time

...@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

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-

[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 fo

Re: [asterisk-users] Delayed answer when calling out

ists.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

[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 __

[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

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: ast

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 infor

[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

[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

[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? ___

[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... -- _

Re: [asterisk-users] Can an agent Login to a queue and be paused

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

Re: [asterisk-users] Can an agent Login to a queue and be paused

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

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-tutor

[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 f

[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 an

[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 anythin

Re: [asterisk-users] Door Phone Assistance

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... > > W

[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

Re: [asterisk-users] Not hearing Telco Operator messages

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."

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.

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] 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, Rober