Re: [asterisk-users] How to add contexts in asterisk realtime?
2008/10/24 Steve Murphy [EMAIL PROTECTED] Well, if you have 50K extensions, you'll find the trunk/1.6.x versions a bit easier to bear in this respect; I've redone the reload process so that it takes longer, but the magic is that it locks the dialplan and swaps in the new dialplan in about 4-10 microseconds. So, no matter the size of the dialplan, literally no interruption to running code takes place... But you'll find that you can only do so many restarts per unit time... That said, I'd still advise using a db if large numbers of non-pattern numbers are what's in the extensions... I've not done benchmarks on speed, but it could be, that if you use the fast pattern matcher, how is this fast pattern matcher enabled ? is it the default pattern matcher in trunk/1.6.x versions ? that the dialplan lookups could be faster than db lookups. If anybody's done any comparisons, let me know... murf ___ -- 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 -- Steve Murphy Software Developer Digium ___ -- 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] fax / t38 gateway
2008/10/23 Brendan Martens [EMAIL PROTECTED] Indeed I am going for pure voip and trying to figure out how to implement t.38, as you suggest. On Oct 23, 2008, at 2:08 AM, Olivier wrote: I think Brendan is asking about endpoints (how to connect fax machines to pure VoIP). Short answer: - you could connect standalone T.38-enabled analog gateways to 1.4, Like what? I'm not familiar with this tech, I googled around a bit but didn't come up with much. I think I just don't know the lingo yet. : ( Could you point out one of these? Linksys PAP2 or 3102 for instance or Patton M-ATA In fact, I would say most analog gateways with FXS port should also support T.38. In this case, your setup would be : ISTP xDSL --- router ---LAN ---Asterisk 1.4 ---LAN ---analog gateway === fax machine As you mentioned, your IP Telephony Service Provider, would have to provide T.30/T.38 conversion so that whenever you're sending or receiving a fax, it would flow in ou or of your network. - with 1.6, you can also use an analog board inside a server and connect fax machines to this board. So basically what you're saying is that to do this (convert the analog to t.38) myself I would still need to have analog coming into my asterisk server (which makes sense, but doesn't help me avoid paying for normal phone lines)... Sounds to me like in this situation t.38 would be purely for getting faxes around on my own asterisk(s) if that became necessary. What I meant is that, instead of using a separate box for connecting your own fax machine, you could use an analog board such as : ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXS board === fax machine Just as previous 1.4 setup, you wouldn't need a separate analog line for faxing. But judging from your question, I would add that it's not common to find an ITSP able to deliver T.38 services (inbound or outbound). And if you want to be able to switch from one provider to another, or simply for simplicity, it's recommended practice to dedicate an analog line to faxing. You setup becomes : ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXO-FXS board === fax machine || PSTN === I should also add that if you're having a single fax machine, maybe you should just connect it to an analog line. Which leads me to my other question again, is there some sort of internet service that will do the analog to t.38 conversion for me and then pass the t.38 on to my asterisk server? In your previous question you said pure VoIP which implied you had found such provider. Here you will some answers : http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38 ___ -- 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 Andrew you mentioned something about sip providers that support t.38? When you say support, do you mean that they have passthrough turned on, or they will actually do an analog t.30 to t.38 conversion for you? That may be what I'm after... If you, or anyone else, know of a provider that does this could you point me in the right direction? Thank you all for your thoughts. Brendan Martens ___ -- 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] problems with some incoming/outgoing calls
On Thu, Oct 23, 2008 at 11:30:29PM +1100, Fernando Serto wrote: Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not so reliable ISP in australia. For incoming calls, I had a Digium TDM410P with 4xFXO modules and HWEC. Because of these problems, i replaced the Digium card with a Sangoma A200D, but it didn't make any difference to the problems. All phones are hooked up to a Netgear PoE switch. Almost forgot to mention that this is not my first Asterisk setup, and in fact it is my 4th, and I used various SIP handsets before, and also different cards (Analog and Digital), so I'm not a total noob. Let's get to the problems... 1) Some incoming calls cannot be picked up Sometimes, incoming calls, coming through the analog card, cannot be picked up. All handsets are set to ring at the same time on incoming calls. and most of the time, calls can be answered on any of the handsets, but maybe 3 or 4 times a day, all handsets will be ringing, and you go to one handset to answer the call, you pick the handset, and it doesn't answer the call, it keeps ringing, then you go to another handset, and still can't pick up, sometimes, you can even try all 4 handsets, and no luck. but, at other times, you can't answer on the first handset, but you can on another, and it is totally random. but people are pretty pissed off for running around to answer a call. and what puzzles me is that you can sit around watching logs for hours, and it won't happen, other times, it happens 3 times in a row. any ideas? Could you please enable 'sip set debug' in the Asterisk CLI and provide a trace? 2) Delay on outgoing calls via SIP People have been saying that when they call people, there's a delay for the call to be answered. For example, caller dials a number, callee answers the ringing phone, but caller is still listening to a ringing tone, and after a few seconds (up to 15 seconds) it sounds like the callee has just answered the call, when in fact, he had already answered a few seconds before. Problem with this is that some callees will hangup before the caller starts talking. These calls are going via pennytel, in australia, which seems to be a pretty good VOIP provider around here, and I've been using it on other setups and never had these issues. Maybe this is a dialplan issue? Any chance you use freepbx? To see if there's a dialplan issue, use: 'core set verbose 3' in the Asterisk CLI. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
On 24 Oct 2008, at 03:57, David Gibbons wrote: Dare I ask why you want to do this? Cheaper than buying an AIM-CUE? And certainly more flexible. ___ -- 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] switching from 1.6.0-beta9 to 1.6.0.1 problems
Hi I've found a solution for what i think is exactly the same problem here: http://bugs.digium.com/view.php?id=13491nbn=6 Regards Enrico Julien Claassen ha scritto: Hello everyone! I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not working. Here's what happens, if I try to call the line: bach P[ 1] -- !! lib: No free channel! P[ 1] -- we have already send Release_complete I haven't changed the configuration fles. Should I change something there? If you need more info, just tell me and I'll provide it, if I can. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Emerging dilema? DID forwarding meets SMS
On Thu, 23 Oct 2008, Karl Fife wrote: We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. It occurrs to me that relaying SMS messages following dialplan logic may become an increasingly common objective. I say the SMS messages 'naturally' disappear but maybe I'm just ignorant to this topic because it has not been important to us in the past. Er, they don't dissapear for me. I send a TXT to a landline, the phone rings and there is a text to speech robot which reads it out to you, or, you can register to not have that happen, and then it sends it to a device which decodes the tones and puts it on the phone display. (And by a similar method you can send TXTs from a landline phone that has the right facilities) If you don't answer, it tries a few more times, or you can call the number and it'll speak it back to you. Don't you have that facility? Maybe it depends on country and telco. Currently we routinely SEND SMS's from Asterisk triggered by other dialplan events. So far we've never needed to RELAY from one DID to another. Are terrestrial carriers even presented with SMS messages? Is anyone using Asterisk to relay SMS messages? The possibilities probably depend on the country you're in.. Gordon ___ -- 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] fax / t38 gateway
Olivier wrote: Linksys PAP2 or 3102 for instance or Patton M-ATA In fact, I would say most analog gateways with FXS port should also support T.38. In this case, your setup would be : That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. Actually T.38 support is far from universal, and a lot of ATAs with support are as buggy as a roache nest Steve ___ -- 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 and Cisco Call Manager Express (CME)
Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that's a different story. And I dare say that it does sound like a fun project to take on. Dave -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2008 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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] Problems with zaptel/ztdummy/asterisk.
Hi, I've managed to build the zaptel modules including ztdummy; ztdummy is installing fine in the modules list and the relevant device structures are present. lsmod | grep ztdummy gives:- ztdummy 5160 0 zaptel186916 1 ztdummy rtc12372 1 ztdummy Where I'm stuck is I am now at a loss as to how to configure my /etc/zaptel.conf and /etc/asterisk/zapatel.conf files:- Zaptel.conf contents are:- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order #Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 (MASTER) # Global data loadzone= us defaultzone = us Zapatel.conf is blank as i don't know what to put there - found plenty of references on how to populate both files for hardware based zaptel cards but very little information on making both config files work with ztdummy. Trying to fire up asterisk with the modules loaded:- [Oct 24 13:36:50] = [Oct 24 13:36:50] == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] Asterisk Event Logger Started /var/log/asterisk/event_log [Oct 24 13:36:50] ERROR[12348]: asterisk.c:3009 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options: 1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support. 2. You only have to load Zaptel drivers if you want to take advantage of Zaptel services. One option is to unload zaptel modules if you don't need them. 3. If you need Zaptel services, you must correctly configure Zaptel. Without the ztdummy module loaded Asterisk is firing up - but then of course I can't run the meetme application... which is rather key. -- Richard Horton Users are like a virus: Each causing a thousand tiny crises until the host finally dies. http://www.solstans.co.uk - Solstans Japanese Bobtails and Norwegian Forest Cats http://www.pbase.com/arimus - My online photogallery ___ -- 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 and Cisco Call Manager Express (CME)
It's definitely just for fun, I wouldn't think to try to implement such as setup for a client unless I were really comfortable with the setup! On Fri, Oct 24, 2008 at 8:36 AM, David Gibbons [EMAIL PROTECTED] wrote: Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that's a different story. And I dare say that it does sound like a fun project to take on. Dave -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2008 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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] prective dialer
Well, there's no harm in _looking_ at it. ram wrote: look at Vicidial ram On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi everybody This is Yavuz YILDIRIM I am software developer.I have a some problems in asterisk. I am using mysql db. Realtime using asterisk modules. On db i am using calling hundred fields for use dial. But i don't know how i can automaticly dial this fields on records numbers. Who can help me asterisk api and others. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] fax / t38 gateway
I'll look into those devices mentioned.
I think that I have one last question... I don't intend to have a
hardware fax machine on our end, I really just want it to get to
asterisk then email it from there. I know this can be done with
hylafax/iaxmodem etc, I actually have gotten that to work
intermittently, but supposing I did find an ITSP with t.38 support,
what would the steps be on the asterisk to receive that fax? I presume
it would just be something like:
exten = _XX.,n,Answer()
exten = _XX.,n,Wait(3)
exten = _XX.,n,Set(TIFFFILE=/var/spool/faxes/incoming-fax.tif)
exten = _XX.,n,ReceiveFAX(${TIFFFILE})
exten = _XX.,n,Set([EMAIL PROTECTED])
exten = _XX.,n,System('mewencode -e ${TIFFILE} | mail -s fax ${EMAIL}')
exten = _XX.,n,System('rm ${TIFFILE}')
That is what I was trying before I realized that it wouldn't work due
to ReceiveFAX() expecting t.38, right? But if it were coming in as t.
38 that is all there would be to it?
Thanks once again for taking the time to answer my questions.
Brendan Martens
On Oct 24, 2008, at 3:22 AM, Olivier wrote:
2008/10/23 Brendan Martens [EMAIL PROTECTED]
Indeed I am going for pure voip and trying to figure out how to
implement t.38, as you suggest.
On Oct 23, 2008, at 2:08 AM, Olivier wrote:
I think Brendan is asking about endpoints (how to connect fax
machines to pure VoIP).
Short answer:
- you could connect standalone T.38-enabled analog gateways to 1.4,
Like what? I'm not familiar with this tech, I googled around a bit but
didn't come up with much. I think I just don't know the lingo yet. :
( Could you point out one of these?
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In this case, your setup would be :
ISTP xDSL --- router ---LAN ---Asterisk 1.4 ---LAN ---
analog gateway === fax machine
As you mentioned, your IP Telephony Service Provider, would have to
provide T.30/T.38 conversion so that whenever you're sending or
receiving a fax, it would flow in ou or of your network.
- with 1.6, you can also use an analog board inside a server and
connect fax machines to this board.
So basically what you're saying is that to do this (convert the analog
to t.38) myself I would still need to have analog coming into my
asterisk server (which makes sense, but doesn't help me avoid paying
for normal phone lines)... Sounds to me like in this situation t.38
would be purely for getting faxes around on my own asterisk(s) if that
became necessary.
What I meant is that, instead of using a separate box for connecting
your own fax machine, you could use an analog board such as :
ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXS board
=== fax machine
Just as previous 1.4 setup, you wouldn't need a separate analog line
for faxing.
But judging from your question, I would add that it's not common to
find an ITSP able to deliver T.38 services (inbound or outbound).
And if you want to be able to switch from one provider to another,
or simply for simplicity, it's recommended practice to dedicate an
analog line to faxing.
You setup becomes :
ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXO-FXS
board === fax machine
|
|
PSTN ===
I should also add that if you're having a single fax machine, maybe
you should just connect it to an analog line.
Which leads me to my other question again, is there some sort of
internet service that will do the analog to t.38 conversion for me and
then pass the t.38 on to my asterisk server?
In your previous question you said pure VoIP which implied you had
found such provider.
Here you will some answers :
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38
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Andrew you mentioned something about sip providers that support t.38?
When you say support, do you mean that they have passthrough turned
on, or they will actually do an analog t.30 to t.38 conversion for
you? That may be what I'm after... If you, or anyone else, know of a
provider that does this could you point me in the right direction?
Thank you all for your thoughts.
Brendan Martens
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Re: [asterisk-users] fax / t38 gateway
Do you have any recommendations for good ones, or, non-buggy ones? Brendan Martens On Oct 24, 2008, at 7:48 AM, Steve Underwood wrote: Olivier wrote: Linksys PAP2 or 3102 for instance or Patton M-ATA In fact, I would say most analog gateways with FXS port should also support T.38. In this case, your setup would be : That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. Actually T.38 support is far from universal, and a lot of ATAs with support are as buggy as a roache nest Steve ___ -- 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] Emerging dilema? DID forwarding meets SMS
Gordon Henderson wrote: On Thu, 23 Oct 2008, Karl Fife wrote: We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. Er, they don't dissapear for me. I send a TXT to a landline, the phone rings and there is a text to speech robot which reads it out to you, or, Don't you have that facility? Maybe it depends on country and telco. Err, Gordon, you must be in a country from the 21st century. North America is just beginning to emerge from the mobile Stone Age. Some people have heard of text messaging but most think you have to pay Blackberry to send emails. I ran into the issues Karl mentions when trying to txt our ISP contact during our office move. Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] fax / t38 gateway
2008/10/24 Steve Underwood [EMAIL PROTECTED] Olivier wrote: Linksys PAP2 or 3102 for instance or Patton M-ATA In fact, I would say most analog gateways with FXS port should also support T.38. In this case, your setup would be : That list rather poorly supports your argument. Yes, you're right : I meant most business analog gateways ... The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. Actually T.38 support is far from universal, and a lot of ATAs with support are as buggy as a roache nest Steve ___ -- 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] fax / t38 gateway
2008/10/24 Brendan Martens [EMAIL PROTECTED] Do you have any recommendations for good ones, or, non-buggy ones? It should be wise to also ask your ITSP as T.38 interop is far from easy ... Would you go with pure-VoIP or would you keep an analog line ? ___ -- 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] Freepbx or Trixbox Presentation
Please does anyone have Freepbx or Trixbox Powerpoint Presentation? Thanks _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- 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] Problems with zaptel/ztdummy/asterisk.
On Fri, Oct 24, 2008 at 01:38:14PM +0100, Richard Horton wrote: Hi, I've managed to build the zaptel modules including ztdummy; ztdummy is installing fine in the modules list and the relevant device structures are present. lsmod | grep ztdummy gives:- ztdummy 5160 0 zaptel186916 1 ztdummy rtc12372 1 ztdummy Where I'm stuck is I am now at a loss as to how to configure my /etc/zaptel.conf and /etc/asterisk/zapatel.conf files:- Zaptel.conf contents are:- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order #Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 (MASTER) # Global data loadzone= us defaultzone = us Zapatel.conf is blank as i don't know what to put there - found plenty of references on how to populate both files for hardware based zaptel cards but very little information on making both config files work with ztdummy. I can't think of a useful reason for you to actually run ztcfg . That file, or a blank file will do fine. Trying to fire up asterisk with the modules loaded:- [Oct 24 13:36:50] = [Oct 24 13:36:50] == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf [Oct 24 13:36:50] Found [Oct 24 13:36:50] Asterisk Event Logger Started /var/log/asterisk/event_log [Oct 24 13:36:50] ERROR[12348]: asterisk.c:3009 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options: 1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support. 2. You only have to load Zaptel drivers if you want to take advantage of Zaptel services. One option is to unload zaptel modules if you don't need them. 3. If you need Zaptel services, you must correctly configure Zaptel. Without the ztdummy module loaded Asterisk is firing up - but then of course I can't run the meetme application... which is rather key. Try running zttest . Does it print anything or is simply hung? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
Re: [asterisk-users] SIP channels seem not to close after call is finished
I've restarted the service and zombie channels were killed. Daniel On Wed, Oct 15, 2008 at 3:29 PM, Steve Murphy [EMAIL PROTECTED] wrote: On Tue, 2008-10-14 at 17:24 -0500, Daniel - Asterisk wrote: Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [EMAIL PROTECTED] asterisk -rx sip show channels |grep 648 Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.25.29648 7c24869b010 00102/0 0x2 (gsm) No Tx: ACK 192.168.25.29648 26e8187a0a4 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 5289c52b77e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 7a6243bc21e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 32bcf3ea3f9 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 21ff7be5355 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 04725bda23e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 2e9a9db559c 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 7fab5e8044d 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 11313fc173a 00102/0 0x0 (nothing) No Tx: CANCEL 2. Asterisk version: 1.4.21.1 These look a lot like the Zombie Channel Bloating Death problems we attacked over the last few weeks. Please see if the latest svn version of 1.4 has these problems still. In high-volume systems, this looked like a huge memory leak that would lead to death by swiftly using up memory, file descriptors, etc. until Asterisk ran out of virtual memory and crashed. There are a couple of code paths, one leaves CANCELED channels lying around, the other BYE'd channels. murf 3. I'm using SIP realtime peers, sip.conf configuration follows: [general] bindport=5060 bindaddr=0.0.0.0 context=default language=es rtcachefriends=yes disallow=all allow=ulaw allow=alaw allow=gsm rtpholdtimeout=300 rtptimeout=300 dtmfmode=rfc2833 videosupport=yes progressinband=yes allowsubscribe=yes subscribecontext=extensiones notifyringing=yes notifyhold= yes limitonpeers= yes Daniel Arohuanca Lagos +51 1 994149553 Lima-Peru ___ -- 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 -- Steve Murphy Software Developer Digium ___ -- 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] Advice on ISDN and Asterisk in the UK
Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? Any help or suggestions would be gratefully appreciated :-) Cheers Phil ___ -- 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] fax / t38 gateway
Olivier wrote: 2008/10/24 Brendan Martens [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Do you have any recommendations for good ones, or, non-buggy ones? Some of or resellers are using 2102 apparently with no issues :) Senad www.bicomsystems.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] Emerging dilema? DID forwarding meets SMS
On Fri, 24 Oct 2008, Drew Gibson wrote: Gordon Henderson wrote: On Thu, 23 Oct 2008, Karl Fife wrote: We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. Er, they don't dissapear for me. I send a TXT to a landline, the phone rings and there is a text to speech robot which reads it out to you, or, Don't you have that facility? Maybe it depends on country and telco. Err, Gordon, you must be in a country from the 21st century. The UK, and while BT do have their faults, they do have some handy features... North America is just beginning to emerge from the mobile Stone Age. Some people have heard of text messaging but most think you have to pay Blackberry to send emails. I'm sorry. Keep banging the rocks together guys... Last time I visited I was frustrated by the lack of TXTability - too many standards, too many carriers not giving you the full service... The weird thing is that if you have a more or less universal TXTing coverage it would literally take off overnight. It did in the UK when the 4 main operators got together and let TXTs pass between then. I think the latest stats are something stupid like over a billion TXTs a week in the UK now... http://uk.gizmodo.com/2007/11/06/one_billion_text_messages_sent.html However, I've just tried with my VoIP carrier and they just vanish. Might drop them an email and ask about it... I ran into the issues Karl mentions when trying to txt our ISP contact during our office move. Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. If not, there's got to be a killer app in there somewhere if you can figure out a revenue generation mechanism... Gordon ___ -- 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] Advice on ISDN and Asterisk in the UK
Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? I recommend the Sangoma A500 with Echo Cancel. The TE410P is not what your are looking for since its for PRI service. What BT is offering you is BRI service. The Sangoma Wiki will tell you what configs need to me made. http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation Andres http://www.neuroredes.com Any help or suggestions would be gratefully appreciated :-) Cheers Phil ___ -- 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] Advice on ISDN and Asterisk in the UK
On Fri, 24 Oct 2008, Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) BT are sounding desperate! (And why go from 4 channels to 6?) Also watch out what they tie you in for - I suspect they'll offer the free installation if you sign up for a 5-year contract. One of my clients was offered that recently... What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Same hardware you currently have, but with an ISDN2 card rather than TDM400. I use the mISDN drivers, and for the most-part they seem OK. Last one I bought was an openVox B200P from Voipon.co.uk. You'd need the B400P for 6 channels (3 ports) Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? Compile/install/load the mISDN drivers and then add in the configs. It's a bit fiddly, but seems to work OK. I did have some compile issues with the latest mISDN drivers though. Or port it into VoIP and wave at BT ... Gordon -- www.drogon.net ___ -- 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] Advice on ISDN and Asterisk in the UK
Andres wrote: Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? I recommend the Sangoma A500 with Echo Cancel. The TE410P is not what your are looking for since its for PRI service. What BT is offering you is BRI service. The Sangoma Wiki will tell you what configs need to me made. http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation He probably got confused with the B410P which is BRI, as far as working well on BT I can't say but I've used it with France Telecom with no problems. DC ___ -- 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] fax / t38 gateway
I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. 2. Why would anyone terminate a FAX call coming through Asterisk in a FAX machine? Isn't there a way to capture it electronically? If so, it seems that putting the electronic documents in a queue where people can open them, save them, and if they wish, print them would be much more useful (and planet friendly, since a lot aren't worth putting on paper). IMHO, there are only three realistic needs: A. Electronic end to end document transfer which is best done with E-Mail and not telephony. B. Receipt of FAX from outside (old school) sources, which is best done electronically. C. Generation of FAX to outside (old school) destination, which could be done either electronically or in the traditional manner. If end to end FAX is desired, is there any reason why Asterisk should treat it any differently than any other call? The FAX machines on each end generate and decode the information, VoIP is simply an audio channel through which is passes. I don't know what T38 defines or implies, but if it is anything other than how to electronically decode a voice call that happens to contain FAX information (rather than passing it on to a real FAX machine) then I'm not sure what use it is. It would seem to me that the OP needs a way to electronically capture calls that turn out to be FAXes. Wilton ___ -- 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] Advice on ISDN and Asterisk in the UK
Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) Wow, you are lucky. I used to have an ISDN-2 line into my home office. BT wrote to me about 2 years ago and said they were discontinuing the service. They converted my dual channel BRI back into a single POTS. I built a little Asterisk server, stuck an X100p in it for backup calls should my broadband go down (on a separate POTS line) and got two non-geo 0844 IAX trunks for free instead. Who lost out there then? Cheers Alan ___ -- 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] fax / t38 gateway
On Oct 24, 2008, at 9:49 AM, Wilton Helm wrote: I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. 2. Why would anyone terminate a FAX call coming through Asterisk in a FAX machine? Isn't there a way to capture it electronically? If so, it seems that putting the electronic documents in a queue where people can open them, save them, and if they wish, print them would be much more useful (and planet friendly, since a lot aren't worth putting on paper). I can answer both of those with a single point. We just switched (entirely) to Asterisk a few weeks ago. We looked, very briefly, at various ways to get rid of the physical, analog, fax machines. They all ended with the answer People can't figure out e-mail as it is, they aren't going to figure out how to fax via e-mail.. What we need is a pure VoIP fax machine. Daniel Wilton ___ -- 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] Problems with zaptel/ztdummy/asterisk.
2008/10/24 Tzafrir Cohen [EMAIL PROTECTED]: On Fri, Oct 24, 2008 at 01:38:14PM +0100, Richard Horton wrote: Try running zttest . Does it print anything or is simply hung? Hangs - I also found after sending my message my syslog filling up with rtc interupt missed messages - don't think my particular configuration will work ztdummy, going to try switching to our production spec kit rather than my dev box on Monday to see if (as I hope) its a hardware related issue. Sorry for disturbing people -- Richard Horton Users are like a virus: Each causing a thousand tiny crises until the host finally dies. http://www.solstans.co.uk - Solstans Japanese Bobtails and Norwegian Forest Cats http://www.pbase.com/arimus - My online photogallery ___ -- 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] Agents log in afterhours
Hi all, I received a report of a client which stated that two of its agents are logging in to the queues when they actually arent there working. They appeared to be logged on all night. They thought they werent logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours? Thanks, Jorge Santiago Alanís Garza Innovación y Desarrollo mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ -- 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] Panasonic x Asterisk ... NO PROBLEM!
Jonn R Taylor wrote: Install a T1 between the Panasonic and Asterisk and program the T1 in the Panasonic as a other custom PBX. VOIP card would be the best. Jonn One thing to beware of with the Panasonic VoIP card, is that I have found no way of getting it to pass out of band DTMF, possibly because it handles this in a proprietary way. This has been my experience with a TDA100 and VoIP card. Regards, Richard ___ -- 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] fax / t38 gateway
People can't figure out e-mail as it is, they aren't going to figure out how to fax via e-mail.. I can understand people saying that. Myself, I'd take E-Mail any day. I've been messing with FAX at various facilities for years, and have found it unreliable, as have most people I talk to. Nobody knows of the FAX actually went through, and if it did, whether the result was readable on the other end, not to mention if the wrong person grabbed it, or accidentally threw it away with some SPAM FAX. Oh well, each to his own. Wilton ___ -- 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] Advice on ISDN and Asterisk in the UK
On Fri, 24 Oct 2008, Alan Lord wrote: Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) Wow, you are lucky. I used to have an ISDN-2 line into my home office. BT wrote to me about 2 years ago and said they were discontinuing the service. They converted my dual channel BRI back into a single POTS. Sure it was ISDN2e and not Home or Business Highway? They killed off an the HH and BG lines some time back and converted them back to POTS. I've no idea why - I'd cancelled my HH line some time before the cut-off date. I built a little Asterisk server, stuck an X100p in it for backup calls should my broadband go down (on a separate POTS line) and got two non-geo 0844 IAX trunks for free instead. Who lost out there then? Well, quite. BT have their good points, but also their stupidly bad points too. They phone me up once a month at present and ask me why I'm not placing any outgoing calls with them. When I try to tell them why, (because I run my own phone company!) because my reply is not in the script, they just hang up on me. Gordon ___ -- 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] OT: Disable Polycom 650 Forward Softkey
I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the Forward softkey just before she enters the Page All keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have customers yelling Hello, Hello, can you hear me over every single Polycom in the building. In retrospect, it was pretty funny. However, now that it has happened three, count 'em, three times, I've got to figure out how to disable that softkey. I've looked through the sip.cfg file and can't seem to figure out what option would remove that softkey. Has anyone ever had to do this? What feature should I disable? TIA Bill ___ -- 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] OT: Disable Polycom 650 Forward Softkey
In your phone configuration file, for all lines: divert divert.fwd.1.enabled = 0 divert.fwd.2.enabled = 0 divert.fwd.3.enabled = 0 divert.fwd.4.enabled = 0 divert.fwd.5.enabled = 0 divert.fwd.6.enabled = 0 / The worst part is this is the same softkey as 'hangup', bad design Polycom! When the remote user hangs up first and you use the softkey to hangup as well, you accidently end up forwarding somewhere (users freak out and hit random keys). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Friday, October 24, 2008 13:12 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Disable Polycom 650 Forward Softkey I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the Forward softkey just before she enters the Page All keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have customers yelling Hello, Hello, can you hear me over every single Polycom in the building. In retrospect, it was pretty funny. However, now that it has happened three, count 'em, three times, I've got to figure out how to disable that softkey. I've looked through the sip.cfg file and can't seem to figure out what option would remove that softkey. Has anyone ever had to do this? What feature should I disable? TIA Bill ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! - I sympathize, especially since a client of mine is facing the same situation. A potential update to their configuration involves exactly what you (Kristian) suggest: layering TLS in-between. I've run SIP/RTP and IAX over openVPN without issue routinely. What worries me is that the problem is not related to SIP awareness, and that some erratic performance by the Sonicwall that is benign in most circumstances manifests as a quality issue when carrying media streams. Seems unlikely, but does anybody have any clarity on this? smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Sonicwall potentially causing long ping timesto SIP phones
From my experience, Sonicwall is a nightmare with SIP if you do not have Enhanced OS. General rules I use: -Do not use SIP transformations (the VOIP tab), these cause random RTP issues, and once you start forwarding calls between users, all things go to heck. You are better off using NAT/qualify in your sip.conf. -Do not use SonicOS Standard (all new Sonicwalls should come with Enhanced now anyway) as there is no method to increase the timeout for UDP rules, this will never be added to this firmware -In SonicOS Enhanced, create inbound and outbound permit rules for all UDP traffic to your PBX (assuming it is on the WAN side), set the UDP timeout to 300 or more, this covers SIP and RTP, but you can be more specific if you prefer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Friday, October 24, 2008 13:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sonicwall potentially causing long ping timesto SIP phones Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! - I sympathize, especially since a client of mine is facing the same situation. A potential update to their configuration involves exactly what you (Kristian) suggest: layering TLS in-between. I've run SIP/RTP and IAX over openVPN without issue routinely. What worries me is that the problem is not related to SIP awareness, and that some erratic performance by the Sonicwall that is benign in most circumstances manifests as a quality issue when carrying media streams. Seems unlikely, but does anybody have any clarity on this? ___ -- 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] OT: Disable Polycom 650 Forward Softkey
On Oct 24, 2008, at 1:12 PM, Bill Andersen wrote: I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the Forward softkey just before she enters the Page All keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have customers yelling Hello, Hello, can you hear me over every single Polycom in the building. In retrospect, it was pretty funny. However, now that it has happened three, count 'em, three times, I've got to figure out how to disable that softkey. I've looked through the sip.cfg file and can't seem to figure out what option would remove that softkey. Has anyone ever had to do this? What feature should I disable? TIA Bill Instead of disabling the keys on the phone, why not just put logic in your dialplan that refuses calls to the paging extension except when the originator is a handset? If the call != handset originated, then send to the voicemail of the handset that bounced the call. You could possibly do this based on caller ID. This keeps the functionality of the forward and page keys, without leading to the unusual circumstances you describe. (and it's good practice to figure out what happens in bogus loop events, anyway - what if someone forwards their handset back to the main number? Or to a number that doesn't exist? It should _probaby_ then go to the voicemail box of the forwarding extension or user.) JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] OT: Disable Polycom 650 Forward Softkey
That did the trick. And yes, I agree it is a very poor design. After looking at how it all transpired, it made more sense as to why it has happened lately. I recently purchased a wireless headset for the receptionist. She would not use her corded headset because she also does some filing and it kept her mobility down. With the wireless headset, she can move around so she will actually use it. As a side effect, she doesn't lift the handset anymore and is now using the Answer and Hangup softkeys. Aaah. So That's why it just all of a sudden started happening. I knew there would be a downside to that wireless headset :) Thanks again. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Friday, October 24, 2008 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey In your phone configuration file, for all lines: divert divert.fwd.1.enabled = 0 divert.fwd.2.enabled = 0 divert.fwd.3.enabled = 0 divert.fwd.4.enabled = 0 divert.fwd.5.enabled = 0 divert.fwd.6.enabled = 0 / The worst part is this is the same softkey as 'hangup', bad design Polycom! When the remote user hangs up first and you use the softkey to hangup as well, you accidently end up forwarding somewhere (users freak out and hit random keys). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Friday, October 24, 2008 13:12 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Disable Polycom 650 Forward Softkey I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the Forward softkey just before she enters the Page All keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have customers yelling Hello, Hello, can you hear me over every single Polycom in the building. In retrospect, it was pretty funny. However, now that it has happened three, count 'em, three times, I've got to figure out how to disable that softkey. I've looked through the sip.cfg file and can't seem to figure out what option would remove that softkey. Has anyone ever had to do this? What feature should I disable? TIA Bill TOP: That did the trick. ___ -- 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] Panasonic x Asterisk ... NO PROBLEM!
Can i install Asterisk beside Nortel PCM, just for recording all calls on E1 (incoming and outgoing calls) I want to get the E1 into Asterisk (Digium) how can this scenario be achieved in details please ? Date: Sat, 25 Oct 2008 07:42:09 +1300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM! Jonn R Taylor wrote: Install a T1 between the Panasonic and Asterisk and program the T1 in the Panasonic as a other custom PBX. VOIP card would be the best. Jonn One thing to beware of with the Panasonic VoIP card, is that I have found no way of getting it to pass out of band DTMF, possibly because it handles this in a proprietary way. This has been my experience with a TDA100 and VoIP card. Regards, Richard ___ -- 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 _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- 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] Emerging dilema? DID forwarding meets SMS
On Oct 24, 2008, at 9:29 AM, Gordon Henderson wrote: On Fri, 24 Oct 2008, Drew Gibson wrote: Gordon Henderson wrote: On Thu, 23 Oct 2008, Karl Fife wrote: We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. Er, they don't dissapear for me. I send a TXT to a landline, the phone rings and there is a text to speech robot which reads it out to you, or, Don't you have that facility? Maybe it depends on country and telco. Err, Gordon, you must be in a country from the 21st century. The UK, and while BT do have their faults, they do have some handy features... North America is just beginning to emerge from the mobile Stone Age. Some people have heard of text messaging but most think you have to pay Blackberry to send emails. I'm sorry. Keep banging the rocks together guys... Last time I visited I was frustrated by the lack of TXTability - too many standards, too many carriers not giving you the full service... The weird thing is that if you have a more or less universal TXTing coverage it would literally take off overnight. It did in the UK when the 4 main operators got together and let TXTs pass between then. I think the latest stats are something stupid like over a billion TXTs a week in the UK now... http://uk.gizmodo.com/2007/11/06/one_billion_text_messages_sent.html However, I've just tried with my VoIP carrier and they just vanish. Might drop them an email and ask about it... I ran into the issues Karl mentions when trying to txt our ISP contact during our office move. Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. If not, there's got to be a killer app in there somewhere if you can figure out a revenue generation mechanism... Gordon You're right, there is revenue there. That's why carriers haven't done it yet - the FUD keeps them from offering the product. Here in North America, we are lucky to even have the two stones to bang together to make calls. Everyone is in love with short codes, which really kind of suck for low-cost, low-friction messaging since not every one of your users can have a short code for inbound messages. But the revenue is there for shortcodes, and mobile carriers are terrified that SMS-enabling ordinary E.164 numbers will take away their death-grip on the mobile messaging market. I'm of the opinion that there is some sort of collusion happening, but I'm so far away from that these days it doesn't bother me other than to laugh at how backwards our mobile carrier market is here. So when I _did_ care about these things, I spent some time researching it. After a lot of painful phone calls asking obvious questions of carriers (You want WHAT?! IMPOSSIBLE!) the only thing I found was this: Level 3 offers SIP-delivered numbers (origination and termination) which can be SMS-enabled. The SMS-enabling requires a separate deal with a company called Syniverse. But once you get both of those deals in place, you could send/receive messages to numbers which were delivered to you via VoIP trunks. The SMS delivery had various different protocols options over which it could be delivered/accepted from your location. This was 1.5 years ago that I did the research on this, so perhaps vendors other than Level 3 are offering this now in the United States. I hope so. But it was new, cutting-edge crazy stuff back then, despite being COMPLETELY OBVIOUS that the market needs something like this, and that every ITSP would offer it immediately. As far as SMS-enabling existing E.164 addresses that you might have - good luck. If someone knows of a way, let me know since I figure it'll be a cold day in hell before my carrier(s) would offer that service capability. Asterisk isn't the greatest platform yet for accepting text messages, and it's only marginally good at sending them on some types of digital circuits. SIP SIMPLE or SMPP are really the primary protocols for this type of transmission, and Asterisk doesn't have either yet. It's a chicken-and-egg thing, I think - as soon as better SMS transmit/ receive is possible, better text message handling will appear in Asterisk (your code is welcome!) Lastly: there is some activity towards SMS support in some unusual configurations from the OpenBTS guys who are building interesting
Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
John Todd wrote: Instead of disabling the keys on the phone, why not just put logic in your dialplan that refuses calls to the paging extension except when the originator is a handset? If the call != handset originated, then send to the voicemail of the handset that bounced the call. You could possibly do this based on caller ID. This keeps the functionality of the forward and page keys, without leading to the unusual circumstances you describe. (and it's good practice to figure out what happens in bogus loop events, anyway - what if someone forwards their handset back to the main number? Or to a number that doesn't exist? It should _probaby_ then go to the voicemail box of the forwarding extension or user.) I might look into doing that for all other extensions, but for the receptionist phone, we will never want to set a fixed forward using the Polycom Forward softkey. I have logic in my dialplan that routes calls should the receiptionist not get to the call in time or the phone is rebooting for whatever reason. Removing the softkey was the quickest way to remedy the problem. All other phones in the building can still use the forward function. Bill ___ -- 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] Fresh installed box
after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Thanks _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- 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] Sporadic One Way Audio
I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Fresh installed box
after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- 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] Asterisk iaxy adapter annoying beep during conversation
When using an Asterisk iaxy adapter every 15 to 30seconds there is a loud annoying beep during conversation. Does anybody know how to stop it? -- #Joseph ___ -- 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] Sporadic One Way Audio
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Sporadic One Way Audio
The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Sporadic One Way Audio
Well, if this is snom specific I can't offer more insight. It really sounds like misconfigured iptables and/or sip helper (conntrack/nat/etc). Are you sure your IP address is right in your sip.conf? If you don;t have NAT set to yes for these phones, they will trust the sip header for IP address and may misroute. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 7:36 PM To: Asterisk Users List Subject: Re: [asterisk-users] Sporadic One Way Audio Importance: High The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Sporadic One Way Audio
We have seen cases where an IP address conflict caused something like this. You can take Wireshark traces on the PC (possibly run them in a loop so that you have a pretty long context) and if you have one-way audio be quick to log on to the web interface of the phone and also take a wireshark (PCAP) trace. There are a couple of tools available that may help to track such problems down: http://manageengine.adventnet.com/products/vqmanager, http://palladion.net, www.networkinstruments.de, and www.voipfuture.com. I know some of them offer a 14-days demo, and it tremendeously helped on of our clients to fix network problems. You can also use SNMP tools to poll if the phone has any blackouts regaring network availbility (see http://wiki.snom.com/SNMP). Also the phone sends a statistics at the end of each call. Check the BYE message, there is a counter of received and transmitted packets. Those numbers should be roughtly the same. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Brent Davidson Gesendet: Freitag, 24. Oktober 2008 18:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Fresh installed box
queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: "Matt Gibson" [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the "attach voicemail" option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- 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] Fresh installed box
http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- 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] OT: Disable Polycom 650 Forward Softkey
If you don't like the forward key why not simply get rid of it. With firmware 3.1.0 all you need to do is add one line to your config file: softkey.feature.forward=0. While you are at it, you might (or might not) like to get rid of buddies and mystatus. softkey.feature.buddies=0 softkey.feature.mystatus=0 Once you have got rid of the default keys, you might wish to replace them with something more useful to you. The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? -crjw- ___ -- 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
