[asterisk-users] Asterisk stops responding

I am having a problem with an Asterisk 1.6.2.15 server that runs a small call center with Queuemetrics. In the past month we've had this problem 3 times. The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to

Re: [asterisk-users] Top Posting

On 11-01-14 07:42 PM, Don Kelly wrote: > Top Posting refers to the practice of sending a message with a reply at the > top and including the entire thread below the reply. I prefer this. If I'm > actively following a thread, the most-recent information appears at the top > of the message I receive.

Re: [asterisk-users] Top Posting

Whatever your preferred style, the following post is at least worth considering. http://brooksreview.net/2011/01/interleaved-email/ My belief is that it would be nearly impossible for me to follow a high volume list if top posting was the preferred style. For example, the following email fro

Re: [asterisk-users] Top Posting

I can agree that the entire signature is "not relevant to [the] list." but I hope you won't find an example of it adding eight lines to every post. I generally try to include it in only one post in case someone wants to get in touch with me. With regard to the trimming and snipping, I'd prefer to

Re: [asterisk-users] Top Posting

On Jan 14, 2011, at 8:52 PM, Don Kelly wrote: I have nothing to add to the nascent flame war that I thought we had so narrowly avoided when I sent my last message. However: > What did you mean, Andrew, about "Don's multiple > signatures which I think he will review"? > > --Don [snip] Andrew m

Re: [asterisk-users] Why are 4 ports used for a single call?

On Jan 14, 2011, at 7:12 PM, Bruce B wrote: > Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it > as well? I am talking strictly in case of Asterisk. Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC. Tom -- __

Re: [asterisk-users] Bruce B

Now boys, play nice! BTW - the word is LOSER, and ( later on ) LOSE. Words and their spelling do matter John Novack Bruce B wrote: LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my

Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2

Magnus, I finally got to testing the patch myself.  Apparently it did not work for me.  That means that if you are affected by the same issue it is not fixed yet.  The current manifestation an incoming OOH_323 call fails in 30 seconds. After reading your in

Re: [asterisk-users] Bruce B

LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my first posts: "I was going to respond with some very insightful and helpful information but I'm not a "PRI Guru". Sorry, maybe next time."

Re: [asterisk-users] Top Posting

Since I don't want anyone bitch at my spelling again: news up = nose up :-) -Bruce On Fri, Jan 14, 2011 at 8:55 PM, Bruce B wrote: > It was only the people who ONLY asked in a response to go to Google to find > answers that annoyed me but slowly posting preference adds up as well. > > As long

Re: [asterisk-users] Top Posting

It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail,

Re: [asterisk-users] Top Posting

Awww...that's no fair. Andrew has bottom-posted to this top-post thread. That really confuses me. Andrew's 'header' appears at the top of the 'stuff,' and his comments at the bottom. Then there's a little ing that I would have considered really important when I only had 16KB of memory to work wit

[asterisk-users] Bruce B

You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find

Re: [asterisk-users] Top Posting

> Seconded.  Although I've succumbed to bottom posting on occasion when > following the convention of the ongoing thread. > > On 01/14/2011 07:42 PM, Don Kelly wrote: > > Bruce et al… > > I’m posting a new thread with the “Top Posting” subject so I won’t draw > complaints about “hijacking” the 4-po

Re: [asterisk-users] AGI->Macro w/Agruments

So, I take it nobody has a solution to this? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Friday, January 07, 2011 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-use

Re: [asterisk-users] Top Posting

Seconded.  Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al…   I’m posting a new thread with

[asterisk-users] Top Posting

Bruce et al. I'm posting a new thread with the "Top Posting" subject so I won't draw complaints about "hijacking" the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm activel

Re: [asterisk-users] Why are 4 ports used for a single call?

On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes wrote: > On Jan 14, 2011, at 6:45 PM, Bruce B wrote: > > > You really want to read the LONG LONG signature from some people before > you read the actual latest message? I don't know about thatI guess it's > a preference. > > Suffice it to say, Bruce,

Re: [asterisk-users] Why are 4 ports used for a single call?

On Jan 14, 2011, at 6:45 PM, Bruce B wrote: > You really want to read the LONG LONG signature from some people before you > read the actual latest message? I don't know about thatI guess it's a > preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundre

[asterisk-users] Tools to Monitor Asterisk Servers and VMs

Hi Everyone, Are there any generally accepted and widely used tools made and tailored to be used for purpose of monitoring Asterisk servers? I am wondering if there is anything that the Asterisk community mostly uses or are there lots of manual scripts written and nothing really exists that every

Re: [asterisk-users] Why are 4 ports used for a single call?

You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Back to my other questions, now that UDP is clear for me, what ports does SIP require? TCP/UDP 5060 ? and why are there recommendations

Re: [asterisk-users] Why are 4 ports used for a single call?

On Jan 14, 2011, at 5:24 PM, Bruce B wrote: > So, simply pressing Reply and typing in the first line (using gmail webmail > without any clients) is a sin here? How is that top posting??? probably your > clients reading that way? It may be a sin here, but it is certainly impolite many place

Re: [asterisk-users] Selecing the E1 cards for the call center

On Fri, 14 Jan 2011, Tom Rymes wrote: While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) What 'rymes' with flame bait? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.co

Re: [asterisk-users] Ghost ringing

Hey Mike, Is there any chance that you still have this firmware around or can get it? It's unavailable through the Polycom site and through their support. Thanks much in advance, Joe > It`s possible the firmware problem is caused by higher (or lower?) > latencies. I can only report on my own exp

[asterisk-users] logging

We have queuemetrics, qloaderd and mysql running on our asterisk server in order to streamline call reporting. Now in order to get internal call logs, I need to get thirdlane Master.csv file. I see there is an option in thirdlane to import this into mysql which would make it easier to work with a

Re: [asterisk-users] Why are 4 ports used for a single call?

On Fri, 14 Jan 2011 17:29:26 -0500, Tom Rymes wrote: >On 01/14/2011 4:19 PM, Bruce B wrote: >> Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 >> right? and why are there recommendations of opening 5000-5082 UDP for >> SIP along with 5060 TCP? Are there any "niceties" to that a

Re: [asterisk-users] Spectralink 8002

The traffic I originally posted was all (except the DHCP request/response) that the phone did since power on. That was sniffed at the output of the wireless controller (all APs tunnel back to the controller). The wireless controller shows the phone as "connected", but I haven't gone much further

Re: [asterisk-users] Why are 4 ports used for a single call?

So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? On Fri, Jan 14, 2011 at 5:13 PM, Tilghman Lesher wrote: > On Friday 14 January 2011 15:12:29 Bruce B wrote: > > Of

Re: [asterisk-users] Why are 4 ports used for a single call?

On 01/14/2011 4:19 PM, Bruce B wrote: Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any "niceties" to that as well? maybe video transmission stuff? More likely, it's bec

Re: [asterisk-users] Spectralink 8002

Do you see any attempt on the wireless controller from the phone to connect to anything on the network after the TFTP exchange? Any traffic at all on the network from the phone? Have you tried to capture packets with Wireshark or something similar? On Fri, Jan 14, 2011 at 4:15 PM, Jonathan C. Ba

Re: [asterisk-users] Spectralink 8002

It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: "MrHanMan" To: "Asterisk Users Mailing List

Re: [asterisk-users] Why are 4 ports used for a single call?

On Friday 14 January 2011 15:12:29 Bruce B wrote: > Off topic - what is top post? I am using gmail + chrome - no ugly > Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiq

Re: [asterisk-users] Spectralink 8002

The only time I've seen "no net found" on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time

Re: [asterisk-users] Spectralink 8002

I figured that (since the firmware is "current" on the phone). I just can't figure why it won't connect. I did notice the phone was showing "no net found" and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: "MrHanMan" To: "Asterisk Users Mailing List

Re: [asterisk-users] Spectralink 8002

If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an er

Re: [asterisk-users] Why are 4 ports used for a single call?

Got it. Thanks. Makes sense to keep an extra two in mind for conference etc Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas wrote: > Hurray for Microsoft Outlook (for creating this whole top-post thread). > Just my

Re: [asterisk-users] Why are 4 ports used for a single call?

Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any "niceties" to that as well? maybe video transmission stuff? Thanks again, On Fri, Jan 14, 2011 at 4:12 PM, Bruce B wrote

Re: [asterisk-users] Asterisk 1.8.2 Now Available (not 1.8.3)

Sorry, subject is incorrect. The released version is Asterisk 1.8.2. On 11-01-14 03:12 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/

[asterisk-users] Spectralink 8002

Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wi

Re: [asterisk-users] Why are 4 ports used for a single call?

Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports - I think that the 4 port requirement is probably for "niceties" like conferencing and transfers. _ From: asterisk-

Re: [asterisk-users] Why are 4 ports used for a single call?

Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So,

[asterisk-users] Asterisk 1.8.3 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible wit

[asterisk-users] Asterisk 1.6.2.16 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.2.16. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.16 resolves several issues reported by the community and would have not been possib

[asterisk-users] Asterisk 1.4.39 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible w

Re: [asterisk-users] Selecing the E1 cards for the call center

Tom Rymes wrote: While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) Many thanks, Tom That is certainly a religious argument that will NEVER have one right answer. vi is probably found on most systems, but that is certainly no reason to use it! Joh

Re: [asterisk-users] Why are 4 ports used for a single call?

RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B wrote: > I mean part of R

Re: [asterisk-users] Why are 4 ports used for a single call?

Each media stream will use two, one for RTP and one for RTCP. In your case 10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is always and even numbered port, and RTCP is always RTP port + 1. Yes, it's in the RFC for RTP. The fact that you have two pairs means that two

Re: [asterisk-users] Selecing the E1 cards for the call center

While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) Many thanks, Tom PS: Bilal: You have asked a nearly unanswerable question. Some prefer one, some prefer the other. Both cards are quality items. I can say that I only have experience with Sangoma T1/E1 c

Re: [asterisk-users] Why are 4 ports used for a single call?

I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B wrote: > Hi Everyone, > > I am just tweaking a pfSense router and learning lots about NAT etcI > noticed that each call uses four UDP port for RTP. Here is an example of > port for a call I made: > > 10200 > 10201 > 10504 > 105

[asterisk-users] Why are 4 ports used for a single call?

Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so

Re: [asterisk-users] Ghost ringing

It`s possible the firmware problem is caused by higher (or lower?) latencies. I can only report on my own experience, which is peace and quiet since I switch (I think the good version to have with respect to that issue was 3.3.0) Mike -Original Message- From: asterisk-users-boun...@lists.

Re: [asterisk-users] Ghost ringing

Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still experiencing issues; the office isn't. If it were firmware, I'd assume the issue would 'travel' with the phone. Though I can give re-flashing it a shot.

Re: [asterisk-users] Ghost ringing

I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Mes

Re: [asterisk-users] Ghost ringing

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the

[asterisk-users] Selecing the E1 cards for the call center

Hi All; We would like to build a call center having 2 E1, but we would like to know which card to select: Sangoma or Digium? And card type to be PCI express or PCI 5.0V or PCI 3.3V ? Any advise or special recommendations for the call center? Regards Bilal --

Re: [asterisk-users] CallerID and URL pop up for windows...

I tried a lot of these softwares in the past few days and lots of them are just a pile of .. lots of compatibility issues with various versions of Outlook and Windows or simply don't do either of inbound or outbound. However, I have been testing Ingeniussoftware and their product so far works w

Re: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at m

Re: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls

On Fri, 14 Jan 2011, Danny Nicholas wrote: I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the c

Re: [asterisk-users] Blind Transfer not working - 1.4.38

Hi, 1.6.2.16rc1 does not have this problem (that`s why I am running a release candidate right now). Can`t say about 1.4 versions, but it`s safe to say whatever they fixed will be out in the next version. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:aster

Re: [asterisk-users] Blind Transfer not working - 1.4.38

This is a heads up to everyone Apparently this is a known but in the latest version on asterisk 1.4, 1.6 and 1.8 http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1 https://issues.asterisk.org/view.php?id=18185 On Thu, 2011-01-06 at 13:10 +, Ishfa

Re: [asterisk-users] CallerID and URL pop up for windows...

On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B wrote: >But better thing >would be to a have TAPI for outlook to query Outlook contact as well because >it allows for making notes on the contact. I am willing to pay for that if >it is added to URAN

Re: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ftarz Sent: Thursday, January 13, 2011 9:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls I'm running Aster

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On Fri, Jan 14, 2011 at 10:46 AM, Jonas Kellens wrote: > On 01/14/2011 02:40 PM, Andrew Latham wrote: >> >> On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellens >>  wrote: >> >>> >>> Hello list, >>> >>> today I experienced the following for the first time : >>> >>> [Jan 14 11:26:18] DEBUG[27654] channel

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 02:40 PM, Andrew Latham wrote: On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellens wrote: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellens wrote: > Hello list, > > today I experienced the following for the first time : > > [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel > '0x114af2c0' > [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel > '0x1

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Hi, We had the same problems. These problems accours when we try to send (from different servers) a lot of IAX calls to one server. (see couple of 100 calls at the same time) When we upgraded asterisk to version 1.8 we didn't get these problems. Regards, Arjan Kroon Van: asterisk-users-boun..

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 02:22 PM, Thorsten Göllner wrote: Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the followin

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoidin

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:1

[asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlo

[asterisk-users] Asterisk+h324m gateway issue

Hi , i worked with h324m gateway for 3g video calling .It  configured successfully . my code in extensions.conf is [from-zaptel] exten => _X.,1,h324m_gw(0@mainmenu) exten=>_X.,n,Hangup [mainmenu] exten => 0,1,h324m_gw_answer() exten => 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video