Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-28 Thread James Lamanna
On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com

Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-26 Thread James Lamanna
On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote: Also, how big does the cache in frame.c grow to? I've recompiled

Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-25 Thread James Lamanna
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote: Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx memory show summary

Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-24 Thread James Lamanna
of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been running now for around 2 days. I've

Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-24 Thread James Lamanna
hypertreading and cut your ram usage to half of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been

[asterisk-users] Size of frame.c cache in Asterisk 11?

2014-11-24 Thread James Lamanna
(Starting a new email topic for this specific issue) Hi, What is the maximum size of the frame.c cache in Asterisk 11 and why does it constantly increase? This is what I'm up to already: $ asterisk -rx memory show summary 3667584471 bytes (3667366799 cache) in4846685 allocations in file

[asterisk-users] High resident memory with 11.14.0 ?

2014-11-21 Thread James Lamanna
Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S

Re: [asterisk-users] High resident memory with 11.14.0 ?

2014-11-21 Thread James Lamanna
Its up to 5.8G of resident memory with 28321 calls processed. The OOM killer is going to kill this soon at this rate (8GB RAM machine). This seems like a pretty serious problem. It looks like I'll need to restart asterisk every night On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama

Re: [asterisk-users] Setting channel musicclass from AGI

2014-10-06 Thread James Lamanna
wrote: On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna jlama...@gmail.com wrote: Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Actually, you can. Any time you can

[asterisk-users] Setting channel musicclass from AGI

2014-10-05 Thread James Lamanna
Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Duplicated DTMF issues

2013-07-05 Thread James Lamanna
Hi, I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B) through SIP. The 1.4.32 box is then connected to the PSTN through PRIs. I've noticed there are occasions where I am seeing duplicated DTMF. I've verified from the SIP trace from the phone that there is only a single

[asterisk-users] Voicemail Prepend not working properly on 1.8.18

2013-04-11 Thread James Lamanna
Hi, I have a problem with forwarding a voicemail and prepending a message to it. If a user just forwards a voicemail, everything works fine. However, if a user prepends a message to the voicemail when forwarding, the voicemail that is forwarded only contains the prepended message and not the

Re: [asterisk-users] DTMF Blips at end of Record() - 1.8.18

2013-02-22 Thread James Lamanna
On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna jlama...@gmail.com wrote: Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF

[asterisk-users] DTMF Blips at end of Record() - 1.8.18

2013-02-20 Thread James Lamanna
Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James --

Re: [asterisk-users] Congestion() forcing PRI channels to be not available

2012-12-20 Thread James Lamanna
/Asterisk+Versions You might get more help or better behavior by updating to a newer more current version of Asterisk, such as 1.8 which will be receiving bug fixes into October 2014. On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.comwrote: Hi, I have a PSTN Asterisk box

[asterisk-users] Congestion() forcing PRI channels to be not available

2012-12-19 Thread James Lamanna
Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the

[asterisk-users] Static on calls - v1.8.15.0

2012-11-08 Thread James Lamanna
Hi, I'm testing out a server with asterisk 1.8.15.0 on it. I'm experiencing static occurring on almost 90% of calls on this particular server. All test phones are using SIP, and calls to/from PSTN servers are delivered using IAX2. I have other production servers running 1.4.x that do not have

Re: [asterisk-users] Static on calls - v1.8.15.0

2012-11-08 Thread James Lamanna
On Thu, Nov 8, 2012 at 8:47 AM, Richard Mudgett rmudg...@digium.com wrote: I'm testing out a server with asterisk 1.8.15.0 on it. I'm experiencing static occurring on almost 90% of calls on this particular server. All test phones are using SIP, and calls to/from PSTN servers are

Re: [asterisk-users] Forcing SIP trunk matching order?

2012-07-10 Thread James Lamanna
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote: No. This is probably because you are using phone numbers as names of devices with type=friend in sip.conf. That's generally a bad idea. The SIP channel matches an incoming call this way: 1. Take the From: user

[asterisk-users] VoIP Company looking for Asterisk/VoIP Engineer

2012-06-29 Thread James Lamanna
Hi, I work for a VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk/VoIP to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin tasks - Devise and implement scalability

[asterisk-users] Restart single dahdi span

2012-04-25 Thread James Lamanna
Hi, Is it possible yet to restart a single Dahdi span in any version of Asterisk? (instead of all of them) Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

[asterisk-users] Asterisk 1.4.42 NOTIFY replies ignore NAT setting

2011-12-30 Thread James Lamanna
Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back

Re: [asterisk-users] Asterisk 1.4.42 NOTIFY replies ignore NAT setting

2011-12-30 Thread James Lamanna
On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user

Re: [asterisk-users] Asterisk 1.4.42 NOTIFY replies ignore NAT setting

2011-12-30 Thread James Lamanna
On Fri, Dec 30, 2011 at 8:35 AM, James Lamanna jlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't

Re: [asterisk-users] Asterisk 1.4.42 NOTIFY replies ignore NAT setting

2011-12-30 Thread James Lamanna
On Fri, Dec 30, 2011 at 11:55 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 12:29 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 8:35 AM, James Lamannajlama...@gmail.com  wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Flemingkpflem...@digium.com  wrote: On 12/30/2011

[asterisk-users] DTMF between sip trunks and PRIs

2011-07-04 Thread James Lamanna
Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008.

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread James Lamanna
Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to

Re: [asterisk-users] DTMF not being heard correctly by far end conference system

2011-01-13 Thread James Lamanna
Hi Duncan, On Wed, Jan 12, 2011 at 10:13 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options. The system is generating audible tones (that I can hear), although I think the audio is

Re: [asterisk-users] Failed SIP registration kicks registered device off?

2011-01-12 Thread James Lamanna
HI Ye, On Mon, Jan 10, 2011 at 10:04 AM, Ye Liu jaux...@gmail.com wrote: Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register

Re: [asterisk-users] Asterisk replying to wrong port for NOTIFY messages

2011-01-08 Thread James Lamanna
Hi Jeff, On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 5 Jan 2011, James Lamanna wrote: See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Hi James, I'm sure it would be the NAT

[asterisk-users] Asterisk replying to wrong port for NOTIFY messages

2011-01-05 Thread James Lamanna
See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx-

Re: [asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler

2010-12-01 Thread James Lamanna
On Mon, Nov 29, 2010 at 10:02 AM, Shaun Ruffell sruff...@digium.com wrote: On 11/27/2010 11:03 AM, James Lamanna wrote: Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8

[asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler

2010-11-27 Thread James Lamanna
Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel

[asterisk-users] SIP NOTIFY to make linksys/cisco SPA BLF go yellow

2010-10-08 Thread James Lamanna
Hi, I was wondering if anyone stumbled upon the correct event in a sip NOTIFY (from a SUBSCRIBE) to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow? I'm trying to differentiate between On the Phone and DND with the BLF. Thanks. -- James --

[asterisk-users] Polling DND status of a Linksys SPA9xx/5xx phone?

2010-08-17 Thread James Lamanna
Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

2010-06-23 Thread James Lamanna
On Tue, Jun 22, 2010 at 6:33 PM, Ryan Wagoner rswago...@gmail.com wrote: -- The out of dialog support was the trick for 1.6.2.9 since it has support for sending a keep-alive. I have attached a modified version of your patch that worked for me. Do you mind if I attach the modified version of

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

2010-06-23 Thread James Lamanna
On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote: completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of registration problem, though more testing in different environments for

[asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

2010-06-22 Thread James Lamanna
If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs:

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

2010-06-22 Thread James Lamanna
On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

2010-06-22 Thread James Lamanna
On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: If you've used Linksys phones against recent Asterisk 1.4

Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)

2010-06-20 Thread James Lamanna
On Sun, Jun 20, 2010 at 5:42 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response

Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)

2010-06-19 Thread James Lamanna
On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber

[asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)

2010-06-18 Thread James Lamanna
It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off

[asterisk-users] Small VoIP company looking for Asterisk Scalability and Maintenance Engineer

2010-06-03 Thread James Lamanna
Hi, I work for a small VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin tasks - Devise and implement high-availability

[asterisk-users] Extension state can get stuck in 'Ringing' state

2010-05-26 Thread James Lamanna
Hi, I've noticed that if a phone goes UNREACHABLE while it is Ringing, when the phone comes back, Asterisk will not clear the channel that was created, so it still thinks it is in the Ringing state. The only way to clear this is to do a soft hangup on the SIP channel or to restart Asterisk.

Re: [asterisk-users] OT: NAT in SPA922

2010-05-07 Thread James Lamanna
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch

Re: [asterisk-users] OT: NAT in SPA922

2010-05-07 Thread James Lamanna
On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote: On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally

[asterisk-users] Registering a Cisco 7965 on 1.4.26

2010-05-05 Thread James Lamanna
Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set

Re: [asterisk-users] Registering a Cisco 7965 on 1.4.26

2010-05-05 Thread James Lamanna
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Wednesday, May 05, 2010 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-05-01 Thread James Lamanna
It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've used both the Grandstream 286 and the Linksys PAP2T. I have been able to get some limited faxing to work using T30 with a PAP2T. Configuration and provisioning of the Linksys is very easy through either

Re: [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?

2010-04-29 Thread James Lamanna
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup:   PRI                  IAX * PSTN ---* Dialplan I've configured a number on the dialplan server

[asterisk-users] Duplicated DTMF with bridged IAX channels maybe?

2010-04-28 Thread James Lamanna
Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX * PSTN ---* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan

[asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)

2010-04-26 Thread James Lamanna
Hi, After playing around with queues a bunch on 1.4.26.2, I noticed a few things, which the patch below addresses. It addresses: - Callers in position 0 will hear periodic/position announcements at a very different rate than all other callers. -- Announcements while in position 0 could be

Re: [asterisk-users] Repeated: Got SIP response 489 Bad event back from

2010-04-10 Thread James Lamanna
On Sat, Apr 10, 2010 at 6:35 AM, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Hi All, I’ve two asterisk servers on the same LAN, both 1.4, and I keep getting “Got SIP response 489 Bad event back from 192.168.3.10” No idea whats causing it. The only references I can find mentions NATing

Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling

2010-04-10 Thread James Lamanna
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find

Re: [asterisk-users] tones detection

2010-04-10 Thread James Lamanna
Hi Jerry, On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis ge...@pagestation.com wrote: I am looking for something in asterisk that will let me record a wav file  in asterisk (which I know how to do) then some other command (external or dialplan) that would read the wave file and tell me if a

[asterisk-users] Asterisk + DRBD Performance

2010-04-10 Thread James Lamanna
Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? If so, is there a performance issue at all when people are recording voicemails and the like? It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? Thanks.

Re: [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes

2010-04-09 Thread James Lamanna
On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is

[asterisk-users] Busy(20) returns non-zero and exits immediately on IAX channel

2010-04-06 Thread James Lamanna
Hi, I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem when trying to play a Busy tone over a IAX trunk from the PSTN. It seems as though Busy(20) returns non-zero immediately (it does not wait 20s), so the caller never hears the busy tone, but the call just appears to hang up.

Re: [asterisk-users] Problem with Sangoma A104 and euroisdn pri

2010-04-02 Thread James Lamanna
On Thu, Apr 1, 2010 at 6:15 AM, Jaap Winius jwin...@umrk.to wrote: [snip] Besides the above error, I also noticed this:    CLI pri show span 1    Primary D-channel: 16    Status: Provisioned, Down, Active    Switchtype: EuroISDN    Type: CPE    Window Length: 0/7    Sentrej: 0    

[asterisk-users] Exceptionally long voice queue length errors...

2010-04-01 Thread James Lamanna
Hi, I'm seeing a lot of Exceptionally long voice queue length errors in my logs, and then I seem to have a problem where Asterisk will drop the registration for a significant number of phones (they go UNREACHABLE), but then they come back approximately a minute later. Is this some sort of load

[asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

2010-03-29 Thread James Lamanna
Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0 Mar 29

Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

2010-03-29 Thread James Lamanna
SC1425 - Dual, dual-core Xeon Processors. I'm hopefully going to be able to stress test this machine to see if I can make it panic again with the PRI card IRQ isolated to CPU0. If so, I'll see if it does the same thing on the other cores... -- James -- Matt On Mon, Mar 29, 2010 at 8:30 PM, James

Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

2010-03-29 Thread James Lamanna
On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna jlama...@gmail.com wrote: On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote: Dell server by any chance? I have a similar problem with a TE220B in a Dell 1950 III server - i've seen several other people having issues

Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

2010-03-28 Thread James Lamanna
that connection so your NAT/firewall won't just close it. Sorry, should have mentioned that all these phones have qualify=yes and nat=yes in sip.conf. Thanks. -- James On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS

Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread James Lamanna
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards asterisk@sedwards.com 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

[asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

2010-03-27 Thread James Lamanna
Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

2010-03-25 Thread James Lamanna
Zeeshan A Zakaria wrote: On Wed, Mar 24, 2010 at 5:42 PM, James Lamanna jlama...@gmail.com wrote: [snip] The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 So I think it is not your

[asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

2010-03-24 Thread James Lamanna
Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming

[asterisk-users] Hung channel problem with 1.4.26.2

2010-02-19 Thread James Lamanna
Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following: - Make a call to another SIP phone that is an intercom call (Auto-Answer) - For whatever reason, the phone happens to go

Re: [asterisk-users] PRI Problems with 1.6.0.10

2010-02-14 Thread James Lamanna
On Fri, Feb 12, 2010 at 12:54 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely

[asterisk-users] PRI Problems with 1.6.0.10

2010-02-12 Thread James Lamanna
Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely to be long gone. This causes a huge issue because I get a bunch of hangup cause 102s

[asterisk-users] Terminate T.38 to PSTN

2009-12-11 Thread James Lamanna
Hi, Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet? I remember seeing an app_gateway floating around at some point a while ago, but I never had any luck with it. Thanks. -- James ___ -- Bandwidth and Colocation Provided

[asterisk-users] MeetMe thinks DAHDI is missing 1.6.0.10

2009-11-05 Thread James Lamanna
Hi, I've noticed that my MeetMe install seems to think chan_dahdi is missing: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) However, it definitely is since I have 3 PRIs functioning normally :) Is there anything I should check before

[asterisk-users] dahdi dies with No more room in scheduler

2009-10-05 Thread James Lamanna
Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear

[asterisk-users] Busy() returns immediately on IAX trunk

2009-10-02 Thread James Lamanna
Hi, I have two asterisk boxes AB connected together via IAX. Phones register to Asterisk box A, and Asterisk box B is the PSTN connection. When dialing a number from a phone registered to A that DAHDI returns as BUSY, the Busy(20) application returns immediately instead of playing the busy tone.

[asterisk-users] Configuring Parallel SIP Trunks

2009-09-01 Thread James Lamanna
Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer

[asterisk-users] Company in Los Angeles looking for Asterisk Network Administration/Maintenance Engineer

2009-08-26 Thread James Lamanna
. Please email me directly for more details or any questions about the position. Thanks. James Lamanna Warp2Biz, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now

[asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?

2009-08-07 Thread James Lamanna
Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can

Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?

2009-08-07 Thread James Lamanna
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com wrote: Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware

[asterisk-users] Asterisk 1.6 and RFC4235

2009-07-30 Thread James Lamanna
Does Asterisk 1.6 fully support RFC4235? Or is it the same implementation as 1.4? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:

[asterisk-users] Weird audio problem with remote IVRs + DMTF

2009-07-09 Thread James Lamanna
Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in

Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered

2009-06-29 Thread James Lamanna
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show

Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered

2009-06-29 Thread James Lamanna
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? I have tried that. Unfortunately restart when convenient doesn't always seem to actually restart asterisk, presumably because there are stuck calls

[asterisk-users] T38 Fax Gateway for Asterisk 1.6

2009-06-26 Thread James Lamanna
Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6

2009-06-26 Thread James Lamanna
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote: Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection

Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6

2009-06-26 Thread James Lamanna
On Fri, Jun 26, 2009 at 1:31 PM, James Lamannajlama...@gmail.com wrote: The use case is that a customer has a fax machine attached to an ATA. The ATA sends T38 to Asterisk over SIP, then I need to forward that out the PSTN. Got it. I'm saying why not skip the ATA and asterisk, and plug the

Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6

2009-06-26 Thread James Lamanna
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote: Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection

Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered

2009-06-05 Thread James Lamanna
Oliver wrote: How many phones are concerned ? The box currently has about 380 active phone registrations. Thanks. Please CC me directly as well because I'm on digest mode. -- James ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered

2009-06-04 Thread James Lamanna
Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this

[asterisk-users] No response to our critical packet problem

2009-05-22 Thread James Lamanna
Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with no response to our critical packet. Calls to voicemail and internal extensions work fine. I understand that everything points to a

Re: [asterisk-users] No response to our critical packet problem

2009-05-22 Thread James Lamanna
and please CC me, I'm reading in digest mode..) -- James On Fri, May 22, 2009 at 10:36 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s

[asterisk-users] Too many notify events causing Asterisk crash?

2009-03-15 Thread James Lamanna
Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these

[asterisk-users] Asterisk and sip router integration

2009-03-06 Thread James Lamanna
Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't

[asterisk-users] Blind transfer from asterisk dialplan (and problems re-parking a call)

2009-03-02 Thread James Lamanna
Hi, Is there a way to do a blind transfer within an asterisk dialplan (like '##')? The reason I need this (I think) rather than a regular Goto() is that I'm trying to do one-touch parking. I can park a call using one-touch parking and then pick it up again, however if I try to re-park the call, it

[asterisk-users] One way audio after IVR tree

2009-02-07 Thread James Lamanna
Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it

[asterisk-users] Minimum version for asterisk and iaxmodem

2009-02-07 Thread James Lamanna
Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the format for the call is unknown. Does anyone know if there is a minimum version of asterisk that is compatible with

Re: [asterisk-users] Minimum version for asterisk and iaxmodem

2009-02-07 Thread James Lamanna
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote: James Lamanna wrote: Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the format

Re: [asterisk-users] Minimum version for asterisk and iaxmodem

2009-02-07 Thread James Lamanna
On Sat, Feb 7, 2009 at 1:44 PM, James Lamanna jlama...@gmail.com wrote: On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote: James Lamanna wrote: Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when

[asterisk-users] Running asterisk on ARM (TS-7800) 1.4.23.1

2009-02-06 Thread James Lamanna
Hi, I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800). Everything compiles fine, but on startup Asterisk always crashes while loading chan_sip. If chan_sip is removed, it starts up fine, but I really need SIP to work. Any ideas? Thanks. -- James

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