[asterisk-users] asterisknow 1.5 with X100P and TDM400P
Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in the right direction for setting up dahdi on asterisk now? I cant seem to get anything to even show when running dahdi show channels from the CLI.. Obviously I need to make it work in a way that will be compatible with asterisknow and the web gui.. Here are the details of my cards.. [r...@pbx asterisk]# dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 02) 2 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 [r...@pbx asterisk]# dahdi_scan [1] active=yes alarms=OK description=Wildcard X101P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X101P location=PCI Bus 00 Slot 15 basechan=1 totchans=1 irq=11 type=analog port=1,FXO [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 00 Slot 14 basechan=2 totchans=4 irq=11 type=analog port=2,FXS port=3,none port=4,none port=5,none Thanks in advance.. ___ -- 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] asterisknow 1.5 with X100P and TDM400P
WipeOut wrote: Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in the right direction for setting up dahdi on asterisk now? I cant seem to get anything to even show when running dahdi show channels from the CLI.. Obviously I need to make it work in a way that will be compatible with asterisknow and the web gui.. Here are the details of my cards.. [r...@pbx asterisk]# dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 02) 2 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 [r...@pbx asterisk]# dahdi_scan [1] active=yes alarms=OK description=Wildcard X101P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X101P location=PCI Bus 00 Slot 15 basechan=1 totchans=1 irq=11 type=analog port=1,FXO [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 00 Slot 14 basechan=2 totchans=4 irq=11 type=analog port=2,FXS port=3,none port=4,none port=5,none Thanks in advance.. ___ -- 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 I managed to get it working.. Not sure what I did exactly but its working.. :) ___ -- 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] New FXO/FXS interface needed..
Hi, My Asterisk box has sat happily doing its job for years now and never had any real issues.. Unfortunately a lightning storm the other night appears to have damaged the TDM400P (1 FXS and 1 FXO port).. Since this system was put together there seem to heave been a lot of developments in the hardware that can be used with Asterisk.. The main players appear to be Digium (obviously), Sangoma and Openvox.. Now I am not wanting to start a war because I know everyone has their preferences when it comes to these type of things.. My question is simply will the other cards just work like the TDM400P did?? Will I have to build extra drivers?? am I going to have headaches trying to get the non digium cards working?? I am still running really old versions of Asterisk and Zaptel so I am sure I am going to have to upgrade the whole system now, unless I just get another TDM400P.. :) Thanks for any input.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk under VMWare
Anyone had any experience with an Asterisk server as a VMWare virtual machine? ___ --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 under VMWare
Mike Clark wrote: Michiel van Baak wrote: On 18:51, Tue 23 Oct 07, WipeOut wrote: Anyone had any experience with an Asterisk server as a VMWare virtual machine? We are running multiple sites as a VMWare virtual machine. All of them are voip only, so I have no idea how it works with T1/E1/POTS interface cards, but as a pure voip setup it works great. Our testing has yielded pretty good results. We had 10 simultaneous calls with ulaw and quality was very good. We are pure VOIP also. Excellent.. Thats very positive.. Now to find a UK based IAX trunk outbound call provider.. :) ___ --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 behind Multi-NAT question
Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Multi-NAT question
Thanks Steve.. So its the same as the dual NAT scenario.. :) Steve Totaro wrote: I have tried it with the best result of one way audio after spending a few days doing everything imaginable. This is the only scenario where I suggest using IAX. Thanks, Steve Totaro WipeOut wrote: Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Best and easiest soft phone for my Dad..
WipeOut wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. Thats for all the replies to my question.. I will have to check them all out and see what works best for him.. Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Time Bandit wrote: Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? [7011] type=friend secret=S0m3S3cur3P4ssw0rd qualify=no notransfer=yes [EMAIL PROTECTED] host=dynamic disallow=all allow=ulaw,alaw,gsm context=from-internal callerid=Marc Charbonneau 7011 hth Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best and easiest soft phone for my Dad..
Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Firewall on AsteriskNow
hi, Is it easy to add to the AsteriskNow system? I am looking to use it to replace my older Asterisk box but I put my asterisk box on the internet and restrict access to specific IP addresses with APF firewall.. So it would be nice if I could install APF on AsteriskNow.. Thanks.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisknow or Trixbox?
I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. Has anyone player with both who can give me the rundown of the basic pros and cons to either.. Which will best suit my setup??.. Currently I have 3 inbound numbers that come in over IAX plus one in/out analogue line to an X100P.. I have a DECT phone extension connecting to a TDM400P.. There are three local SIP extension and one in a remote office.. Inbound numbers are routed to a recorded message specific for that number and then set to ring on multiple phones (different combinations depending on the inbound number) and if not answered routed to a VM box.. Caller ID is changed on the inbound calls so that the phone displays will show which number was called rather than the calling number and so we can answer with the correct greeting.. The analogue inbound calls simply ring on all the phones.. All outbound calls are via the analogue line and use a carrier pre selection prefix for cheap outbound calls.. So which would you suggest I use? Both are in beat versions at the moment.. Which is likely to so stable first? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpeedTouch 780WL
Has anyone had any experience with this router?? I am looking to use it because I want to use a DECT phone in conjunction with VoIP and this seems to check all the boxes for Wi-Fi, ADSL and VoIP all at a good price.. I have never used Speedtouch hardware before so any feedback would be great.. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WiFi VoIP Handsets..
Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Most comprehensive management?
Thanks Tom and Justin.. I did think they were separate entities.. I will pull down [EMAIL PROTECTED] and see where I get to from there.. Justin Biggs wrote: Another FYI: The latest [EMAIL PROTECTED] release (2.8) includes FreePBX (sounded like you thought they were seperate entities). It should be able to manage anything a roll-your-own server could. On 5/8/06, *Tom Vile* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It will do these things just fine. FYI [EMAIL PROTECTED] is an ISO that will setup linux, FreePBX and various other things for you, you can of course just download FreePBX and install it by yourself. http://lists.digium.com/mailman/listinfo/asterisk-users -- Justin Biggs Owner, Biggs Computer Consulting [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 740.501.4781 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Most comprehensive management?
I see that [EMAIL PROTECTED] and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either [EMAIL PROTECTED]'s or FreePBX's interfaces be able to manage everything I need? If both would then which would be better suited? If neither would are there any other systems out there? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXY codec support and questions..
Hi..
I have to setup an extension in a remote location that will use a
cordless analog telephone.. I am looking at the IAXY to do this for
me..Basically the data path will be as follows...
[Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset
Since there are two NAT boxes in the path I know SIP won't work.. I also
don't want to move the Asterisk box to the internet side of the NAT box,
not only from the security perspective but also the potential issues
with the already configured SIP phones that connect to it locally..
So as far as getting over the NAT problem the IAXY seems the way to go..
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
Thanks for any suggestions or input on this setup.. Also any reviews on
the IAXY are welcome..
___
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Re: [Asterisk-Users] IAXY codec support and questions..
Kevin P. Fleming wrote: WipeOut wrote: To save bandwidth I would like to stay away from using the G.711 codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in the docs.. No. The IAXy only supports G.711 ulaw/alaw and ADPCM. I don't know what 'docs' you were looking in, but this page: http://www.digium.com/en/docs/S101I/IAXy.pdf clearly states what is supported. Thanks, I was looking at the install guide.. It may have been in there too and I just missed it.. Unfortunately G.711 is not going to help me.. I could still get by using it but the quality may be an issue when there is other traffic on the line.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best budget IP phone at the moment?
Hi, I am looking for a budget IP phone that can use preferably iLBC or GSM codecs.. Suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream - No dialtone in handset after 1.0.6.7 firmware update..
Hi, I'm just trying to setup a Grandstream BT-102 that I had lying around and haven't used for over a year.. I loaded up the 1.0.6.7 firmware and factory reset the config.. I have it working but there is no sound to the handset ear piece.. On speaker phone it works fine in an echo test.. When using the handset you can't hear anything.. The phone was working perfectly last time it was used and before upgrading the firmware I think it was working.. Anyone else have this issue?? or is there a setting that I need to look at?? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I factory reset a Grandstream BT-102
Hi, Just pulled out the BT-102 because I need to use it again, entered in the TFTP server to get the latest firmware so its now in 1.0.6.7 and i now was to factory default the phone and set it up from scratch.. I tried the instructions (copied below this message) from the latest available version of the user guide on the Grandstream site but it didn't appear to work.. Anyone got any idea how to factory reset these phones? Thanks 8 Restore Factory Default Setting Warning: Restore the Factory Default Setting will delete all configuration information of the device. Step one: Find the Mac Address of the device. The Mac address of the device is located on the bottom of the device. It is a 12 digit number. Step two: Encode the Mac address. The encode rule is: 2 is the first letter on the button 2 so its encoding is 2 . A is the second letter on button 2 so its encoding is 22 . B is the third letter on button 2 and its encoding is 222 . C is the fourth letter on button 2 and its encoding is . 3 is the first letter on the button 2 so its encoding is 3 . D is the second letter on button 2 so its encoding is 33 . E is the third letter on button 2 and its encoding is 333 . F is the fourth letter on button 2 and its encoding is . For example, the Mac address is 000b8200e395, User should encode it as 000222820095 . Step three: Access the phone screen menu, then select the -- reset -- with the up or down arrows keys. Step four: Dial in the encode of the Mac address. Once the correct encode Mac address dial in, the device will reboot automatically and restore the factory default setting. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Load
Hi, Thinking about an IVR application and trying to get a handle on the best way to structure it so that the maximum number of concurrent calls can be achieved.. If the voice prompts were stored in a GSM format and were being played out through an IAX trunk that uses GSM compression would asterisk do a decompress/compress on the audio or would it simply pass through the GSM encoding? Obviously if I could eliminate the decompress/compress activity is would make the server far more scalable.. Thanks.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silent IAX calls getting cut off
Dave Brooks wrote: Hi. I'm new here so I hope this is a sensible question/sensible place for it. I have a PSTN to IAX phone number with voipuser.org that I'm using to test an IVR service. The only trouble is that after approximately 40 seconds of silence (e.g. after background playback of a menu prompt) the call gets cut off. Is this a common problem? I've already set the ResponseTimeout to a big number. It doesn't get sent to the 't' extension, it just says Hungup in * I can easily get around this problem when waiting at a menu (e.g. just background some music or repeat the menu after 30 seconds). But I want to use the ControlPlayback function, and I don't see what I can do to stop the call getting hung-up if the user chooses to pause the audio for a while. I don't think it's an issue with voipuser.org as the same happens with our number from voicepulse. Any advice would be appreciated. Regards, David Brooks What versions of Asterisk are you and your ITSP using? I have just recently resolved a call dropping issue which had no reason for happening but an upgrade to the version of Asterisk solved the problem.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advantage of IAX2 to SIP?
Michael Vogel wrote: Hi! Some - few - providers are using IAX2 as a protocol. Most are using SIP. I know that there are advantages of IAX2 regarding multiple connections. But beside this I'm asking myself (and you all) why I should prefer IAX2 when my SIP connection is working. Are there differences in the performance? Bye! Michael Hi, If your SIP setup is working and you are happy with it then thats cool.. If it aint broke don't fix it.. The advantages of IAX are as you said, you can make multiple calls at the same time but also that NAT isn't an issue and AFAIK it doesn't have as much of an overhead as SIP.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio Drops out at Random - one way
Craig Waddington wrote: Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio Drops out at Random - one way
Craig Waddington wrote: I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? If you haven't set absolutetimeout in your dialplan then there will be no reason to have to reset it back to 0.. I tried the debug but didn't see much useful in there.. probably because I don't know what its telling me.. I am using Snom phones but I am almost sure its not the phone to Asterisk leg because I have an analog phone connected directly to the Asterisk box and it also drops calls.. Its almost certainly the IAX leg.. My problem appears to be purely on outgoing but we don't get a huge amount of incoming calls over the IAX and as far as I know we have never dropped an inbound call, but because of the small inbound call volume I wouldn't take it as fact.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEED HELP!!
Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
Jason Williams wrote: On Tue, 23 Nov 2004 13:17:57 +, WipeOut [EMAIL PROTECTED] wrote: Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. I Have not read you last posts but the usual cause for asterisk dropping calls is the busy detection algorithims, Try turning off busy detect or increasing the busy count variables, in zapata.conf Jason Hi Jason, Please take a look at my previous emails, I am dropping calls over the IAX termination leg and need to try and establish the cause.. The email I sent on Saturday to the list had a log file attached and the one from yesterday was trying to find a way to make IAX more tolerant of network errors (lost packets etc..).. Thanks for replying anyway.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
Senad wrote: Hi, Is your asterisk server in DMZ environment? Hi Senad, No, this server that is having the issue is behind a NAT firewall connecting through IAX to a termination provider.. Being IAX I would think NAT is irrelevant.. Also there is no patten to the dropping of calls, a call can last less than 1 min or over 30 min or anything in between.. My mail to the list on Saturday included a log file that was taken when two calls were dropped.. Maybe you are better are seeing exactly what was going on.. Thanks.. Senad Jordanovic Bicom Systems, The complete systems provider www.bicomsystems.com USA 1-212-400-7921 UK 0870 682 782 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of WipeOut Sent: 23 November 2004 14:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] NEED HELP!! Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
Senad wrote:
Hi Senad,
No, this server that is having the issue is behind a NAT firewall
connecting through IAX to a termination provider.. Being IAX I would
think NAT is irrelevant.. Also there is no patten to the dropping of
calls, a call can last less than 1 min or over 30 min or anything in
between..
My mail to the list on Saturday included a log file that was taken when
two calls were dropped.. Maybe you are better are seeing exactly what
was going on..
[Senad Jordanovic]
Do you mind resending that file?
Senad Jordanovic
Bicom Systems, The complete systems provider
www.bicomsystems.com
USA 1-212-400-7921
UK 0870 682 782
Here it is...
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 1 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:05 DEBUG[-1109894224]: Enabled echo cancellation on channel 2
Nov 20 16:52:05 DEBUG[-1109894224]: Made call 5 into trunk call 16384
Nov 20 16:52:05 DEBUG[-1109894224]: Created trunk peer for
'111.222.333.444:4569'
Nov 20 16:52:05 DEBUG[-1109894224]: Expanded trunk '111.222.333.444:4569' to
6400 bytes
Nov 20 16:52:07 DEBUG[-1109894224]: Received AST_CONTROL_PROGRESS on Zap/2-1
Nov 20 16:52:07 DEBUG[-1095144528]: Ooh, voice format changed to 1024
Nov 20 16:52:15 DEBUG[-1109894224]: Took Zap/2-1 off hook
Nov 20 16:53:35 DEBUG[-1090942032]: Setting NAT on RTP to 0
Nov 20 16:53:36 DEBUG[-1090942032]: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Found
Nov 20 16:53:48 DEBUG[-1090942032]: Auto destroying call '[EMAIL PROTECTED]'
Nov 20 16:55:01 DEBUG[-1109894224]: Exception on 17, channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Got event Wink/Flash(3) on channel 2 (index
0)
Nov 20 16:55:01 DEBUG[-1109894224]: Winkflash, index: 0, normal: 17, callwait:
-1, thirdcall: -1
Nov 20 16:55:01 DEBUG[-1109894224]: Already have a dsp on Zap/2-2?
Nov 20 16:55:01 DEBUG[-1109894224]: Swapping 2 and 0
Nov 20 16:55:01 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:04 DEBUG[-995472]: Exception on 17, channel 2
Nov 20 16:55:04 DEBUG[-995472]: Got event On hook(1) on channel 2 (index 0)
Nov 20 16:55:04 DEBUG[-995472]: Last flash was 2764 ms ago
Nov 20 16:55:04 DEBUG[-995472]: Swapping 2 and 0
Nov 20 16:55:05 DEBUG[-995472]: disabled echo cancellation on channel 2
Nov 20 16:55:05 DEBUG[-995472]: waitfordigit returned 0...
Nov 20 16:55:05 DEBUG[-995472]: Hangup: channel: 2 index = 2, normal = 17,
callwait = -1, thirdcall = 21
Nov 20 16:55:05 DEBUG[-995472]: Released sub 2 of channel 2
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 2, ts=160005,
seqno=47)
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 11, ts=160008,
seqno=48)
Nov 20 16:55:06 DEBUG[-1109894224]: Didn't get a frame from channel:
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: Bridge stops bridging channels Zap/2-1 and
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: We're hanging up IAX2/provider-out/16384
now...
Nov 20 16:55:06 DEBUG[-1109894224]: Exiting with DIALSTATUS=ANSWER.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: inserting a CDR record.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2004-11-20 16:52:05','\Cordless Phone\
1234','1234','90019090909090','local',
'Zap/2-1','IAX2/provider-out/16384','Dial','IAX2/[EMAIL
PROTECTED]/19058289575',181,171,'ANSWERED',3,'Cordless Phone')
Nov 20 16:55:06 DEBUG[-1109894224]: Hangup: channel: 2 index = 0, normal = 17,
callwait = -1, thirdcall = -1
Nov 20 16:55:06 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:06 DEBUG[-1109894224]: Set option TDD MODE, value: OFF(0) on
Zap/2-1
Nov 20 16:55:06 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:08 DEBUG[-1095144528]: Immediately destroying 16384, having
received hangup
Nov 20
Re: [Asterisk-Users] NEED HELP!!
Geoff Nordli wrote: onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. Thanks.. Did you know that you can obtain commercial support for Asterisk? http://www.digium.com/index.php?menu=software_support I am sure it will be cheaper than dropping Asterisk and moving to a different platform. Have a great day! Geoff ___ Hi Geoff, The problem is that I don't think Asterisk is causing the problem (not entirely anyway), I think its the internet and that IAX is too sensitive to packet loss so when the packet loss exceeds a certain threshold it just drops the call instead of trying to recover and maybe having a second or two gap in the voice.. The problem is that a) I can't confirm this because the log isn't clear to me and b) I wouldn't know where to go to try and make IAX more tolerant of lost packets.. My IAX connection runs over an ADSL line and it does get CRC errors and errored seconds occasionally (but these are very low) and obviously the internet is not a guaranteed delivery.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
Eric Wieling wrote: WipeOut wrote: The problem is that I don't think Asterisk is causing the problem (not entirely anyway), I think its the internet and that IAX is too sensitive to packet loss so when the packet loss exceeds a certain threshold it just drops the call instead of trying to recover and maybe having a second or two gap in the voice.. Odd. My Linksys router was rebooting this weekend. Total loss of connectivity, but my IAX2 calls stayed connected. There was about 5 seconds of lost audio, but the call was not dropped. Heck, even the ethernet link light on the Asterisk server went dark and my laptop said the network cable became disconnected. My IRC sessions dropped, my IM connections dropped, my IMAP connections dropped, but my IAX2 calls did not. ___ Hmmm.. that more or less blows out my theory then and puts me back to square one.. :( What codec are you using? Have you got any jitter buffer setting enabled and if so what are they? Have you done anything to the standard IAX settings? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX error tolerence??
Hi, Didn't get any opinions on the log file I mailed onto the list over the weekend so I am continuing to try and track the cause for the dropped calls.. I have a feeling that its to do with IAX being way too sensitive when it comes to packet loss.. Since it is going across the internet it needs to be more tolerant when it comes to errors and packet loss.. Are there any settings (in the conf file or the source) that I can change to make it more able to keep the call connected without dropping it when there are errors even if the sound breaks up for a couple of seconds?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK available SIP phone?
Mike Dent wrote: Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike If you are looking for the cheapest option then I think the Grandstream is still the cheapest phone I know of that works with Asterisk.. You can get them from http://www.voiptalk.co.uk/.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can anyone shed some light on wht these calls were dropped?
Hi,
I need help finding why my system is dropping calls..
I enabled debugging on my box in the hope it would lead me to the answer
as to why my system is dropping calls but unfortunately nothing is
jumping out at me..
I have attached the portion of the messages file for two calls that were
dropped.. (numbers names and ip's have been changes to protect the innocent)
Can someone give me a reason why the calls were dropped? or maybe tell
me where to look
My only thought is that its a loosing some packets as it goes across the
internet to the IAX provider and that the tolerance for packet loss is
set really low so it drops the call completely..
Any help appreciated..
Thanks..
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 1 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:05 DEBUG[-1109894224]: Enabled echo cancellation on channel 2
Nov 20 16:52:05 DEBUG[-1109894224]: Made call 5 into trunk call 16384
Nov 20 16:52:05 DEBUG[-1109894224]: Created trunk peer for
'111.222.333.444:4569'
Nov 20 16:52:05 DEBUG[-1109894224]: Expanded trunk '111.222.333.444:4569' to
6400 bytes
Nov 20 16:52:07 DEBUG[-1109894224]: Received AST_CONTROL_PROGRESS on Zap/2-1
Nov 20 16:52:07 DEBUG[-1095144528]: Ooh, voice format changed to 1024
Nov 20 16:52:15 DEBUG[-1109894224]: Took Zap/2-1 off hook
Nov 20 16:53:35 DEBUG[-1090942032]: Setting NAT on RTP to 0
Nov 20 16:53:36 DEBUG[-1090942032]: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Found
Nov 20 16:53:48 DEBUG[-1090942032]: Auto destroying call '[EMAIL PROTECTED]'
Nov 20 16:55:01 DEBUG[-1109894224]: Exception on 17, channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Got event Wink/Flash(3) on channel 2 (index
0)
Nov 20 16:55:01 DEBUG[-1109894224]: Winkflash, index: 0, normal: 17, callwait:
-1, thirdcall: -1
Nov 20 16:55:01 DEBUG[-1109894224]: Already have a dsp on Zap/2-2?
Nov 20 16:55:01 DEBUG[-1109894224]: Swapping 2 and 0
Nov 20 16:55:01 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:04 DEBUG[-995472]: Exception on 17, channel 2
Nov 20 16:55:04 DEBUG[-995472]: Got event On hook(1) on channel 2 (index 0)
Nov 20 16:55:04 DEBUG[-995472]: Last flash was 2764 ms ago
Nov 20 16:55:04 DEBUG[-995472]: Swapping 2 and 0
Nov 20 16:55:05 DEBUG[-995472]: disabled echo cancellation on channel 2
Nov 20 16:55:05 DEBUG[-995472]: waitfordigit returned 0...
Nov 20 16:55:05 DEBUG[-995472]: Hangup: channel: 2 index = 2, normal = 17,
callwait = -1, thirdcall = 21
Nov 20 16:55:05 DEBUG[-995472]: Released sub 2 of channel 2
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 2, ts=160005,
seqno=47)
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 11, ts=160008,
seqno=48)
Nov 20 16:55:06 DEBUG[-1109894224]: Didn't get a frame from channel:
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: Bridge stops bridging channels Zap/2-1 and
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: We're hanging up IAX2/provider-out/16384
now...
Nov 20 16:55:06 DEBUG[-1109894224]: Exiting with DIALSTATUS=ANSWER.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: inserting a CDR record.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2004-11-20 16:52:05','\Cordless Phone\
1234','1234','90019090909090','local',
'Zap/2-1','IAX2/provider-out/16384','Dial','IAX2/[EMAIL
PROTECTED]/19058289575',181,171,'ANSWERED',3,'Cordless Phone')
Nov 20 16:55:06 DEBUG[-1109894224]: Hangup: channel: 2 index = 0, normal = 17,
callwait = -1, thirdcall = -1
Nov 20 16:55:06 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:06 DEBUG[-1109894224]: Set option TDD MODE, value: OFF(0) on
Zap/2-1
Nov 20 16:55:06 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:08 DEBUG[-1095144528]: Immediately destroying 16384,
[Asterisk-Users] Zaptel init script
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel init script
[EMAIL PROTECTED] wrote: I had problems with the init script not working ing FC2 also. I fixed it by editing the init script and changing 'insmod' to 'modprobe'. Don't know if that will fix your problem or not, but it's worth a try. -- Jim Dossey Computer Services Hi Jim, Thanks for that, it has solved the problem.. -- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel init script
John Millican wrote: -- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. I belive I have seen on the list where wcfxs has been changed to wctdm this may be your problem? John Millican --- I will have to look into that, is appears to still be loading wcfxs according to the init script but maybe it hasn't been updated yet.. I guess thats the problem with being away from this for a while, things change so fast.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 supported?
9. When the hardware checker finds a new Tiger Jet device, just ignore it. (Anyone know how to make it stop bothering me?) Choose Do nothing and it should stop bothering you.. thanks for the install tops I may be having a go with FC3 in the near future.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Find out the reason for dropped calls?
Hi, Is there any method to log the reason a call was ended / terminated / dropped?? I am getting a fairly high nimber of calls being dropped but have no way of working out why.. I need to still upgrade Asterisk to ver 1.0 but I still need a way to track the reason for the call dropping so that I can eliminate Asterisk as the cause or focus on it.. Obviously there are amny companents to my whole system so if I can eliminate any of them its a start to finding the cause.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P noise on ADSL line.
Following on from the message below I have discovered that the X100P causes the SNR on my ADSL line to drop even with the Asterisk box **switched off** and the power unplugged... This seems very strange.. Why should a card in a switched off PC cause noise on a line meaning that it drops out and has to reconnect quite often.. Anyone got any other ideas to try and stop it messing up my internet connection cos its causing havoc with my VoIP calls coming in and going out over the ADSL line.. Later.. WipeOut wrote: Hi, This may be one for the broadband guru's out there.. I have a single analog line coming into the house.. This line is for my ADSL and home phone.. My Asterisk box uses an X100P card to connect to the analog line.. I have a microfilter on the line etc.. The rest of my phone system works inbound and outbound calls via a VoIP provider over the ADSL line.. The problem I am having is that the X100P seems to introduce a lot of noise on the line when it its connected to the phone socket on the microfilter and this causes the ADSL quality to drop quite badly.. When the X100P is not connected I have a signal to noise ratio of 29dB downstream and 30dB upstream (this stays the same when I connect an analog phone) when I connect the X100P the SNR drops to 12dB downstream and 30dB upstream.. At 12dB I get a large number of CRC errors and errored seconds on the ADSL connection.. Anyone got any ideas why the X100P would cause this kind of deterioration? Only thing I can think of is possibly something to do with ring detection or that its acting on some of the frequencies that are being used by the ADSL.. Thanks for any thoughts.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P noise on ADSL line.
Hi, This may be one for the broadband guru's out there.. I have a single analog line coming into the house.. This line is for my ADSL and home phone.. My Asterisk box uses an X100P card to connect to the analog line.. I have a microfilter on the line etc.. The rest of my phone system works inbound and outbound calls via a VoIP provider over the ADSL line.. The problem I am having is that the X100P seems to introduce a lot of noise on the line when it its connected to the phone socket on the microfilter and this causes the ADSL quality to drop quite badly.. When the X100P is not connected I have a signal to noise ratio of 29dB downstream and 30dB upstream (this stays the same when I connect an analog phone) when I connect the X100P the SNR drops to 12dB downstream and 30dB upstream.. At 12dB I get a large number of CRC errors and errored seconds on the ADSL connection.. Anyone got any ideas why the X100P would cause this kind of deterioration? Only thing I can think of is possibly something to do with ring detection or that its acting on some of the frequencies that are being used by the ADSL.. Thanks for any thoughts.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P noise on ADSL line.
Stewart Nelson wrote: I have a single analog line coming into the house.. This line is for my ADSL and home phone.. My Asterisk box uses an X100P card to connect to the analog line.. I have a microfilter on the line etc.. The rest of my phone system works inbound and outbound calls via a VoIP provider over the ADSL line.. The problem I am having is that the X100P seems to introduce a lot of noise on the line when it its connected to the phone socket on the microfilter and this causes the ADSL quality to drop quite badly.. When the X100P is not connected I have a signal to noise ratio of 29dB downstream and 30dB upstream (this stays the same when I connect an analog phone) when I connect the X100P the SNR drops to 12dB downstream and 30dB upstream.. At 12dB I get a large number of CRC errors and errored seconds on the ADSL connection.. Anyone got any ideas why the X100P would cause this kind of deterioration? I suspect that it's not the X100P, but noise from your PC's power supply or motherboard. 1. With the X100P connected, cycle power to the ADSL modem so it renegotiates. If the noise is relatively narrowband, the noisy bins will be avoided and the SNR should improve. 2. Try two filters in cascade. Plug the new filter into the phone socket of the existing filter, and the X100P into the phone socket of the new one. 3. Try a long (25-foot or 50-foot) cord between the filter and the X100P. Try winding the cord into a coil about 8 inches in diameter. 4. Try a ferrite clamp-on core, such as those used to suppress noise in car stereo systems, around the cord from the X100P. If the core has a sufficiently large opening, make a two or three turn coil with the phone cord. 5. Try putting the X100P in a different PCI slot, so it is as far as possible from noisy boards such as video. Good luck, Stewart Thanks for the suggestions.. I have tried another microfilter, the long cable and the cascaded microfilter and all made no difference at all.. I dont think it is the microfilter or the internal house cabling.. Also the fact that a standard analog phone doesn't do it also points to the X100P.. I can't move the X100P to another PCI slot because I only have two in this PC and the other has a TDM400P.. The only thing I could do is setup a completely new PC with Asterisk and go from there.. Guess this means that as usual I have bumped into a problem that no one else has (of knows they have :) ).. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 200 updates
I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 updates
Bastian Schern wrote: WipeOut schrieb: I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? Have a look here: http://www.snom.com/download/share/ Regards Bastian There are some really new versions there... So why is Snom not automatically distributing them anymore? I really liked it when the phone told me there was an update and I just had to press a button.. Now I have to go and find out if there is an update and then setup a server and load it myself.. :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Firmware
hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? What is the current stable firmware version? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Firmware
Duane wrote: WipeOut wrote: hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? Can get it off the web: http://hellofone.com/downloads/ What is the current stable firmware version? 1.0.5.11 Do the ATA's and the phones use the same firmware? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Making asterisk distributed
Trilogy India wrote: Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Thanks Varun This has been discussed a number of times in the past, searching the archives will reveal the various technical reasons why a clustered SIP solution is very difficult to implement.. Your only real option for scalability is to distribute your users over many servers and interlink them with IAX and your dial plan.. For those who don't know how to search the archives.. Goto the bottom of the page at http://www.digium.com/index.php?menu=mailing_list and search.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linux kernel 2.6.6
Dorian Gray wrote: Kevin Walsh wrote: Leif Madsen [EMAIL PROTECTED] wrote: Also, from what I have been told (and I've tested this by building zaptel, but not any of the other sources) is that you no longer need the sourcecode with the 2.6 kernel. You can create a symlink to: /lib/modules/`uname -r`/build/ Instead of the kernel source code. Apparently that is the correct way to compile now with the 2.6 kernel. Please correct me if I am totally wrong as it is not my intension to give false information (but like I said, I built zaptel against that, and it works great for me). Either way is fine. I build the kernel from source every time (Gentoo), so I always have it as an available symlink target. If you're forced to use a binary kernel distro then symlinking to /lib/modules/... is fine. I believe linking to /lib/modules/xyz/build is the new and approved way that should be done for all external drivers in 2.6, whether it's a binary kernel or from source. (of course, I'm going to continue using /usr/src/linux for a while, heh...) I found linking to /lib/modules/XXX/build far simpler and would build far easier than linking to anywhere else.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1 user 2 VM boxes?
Hi, Would be interestd in anyones ideas for this problem.. We are starting a new division to our company, the people in this new division will be the same people who are on the old division.. Calls for each division come in on seperate numbers and go through seperate menus but ring to common extensions, this is easy enough.. The problem is with Voicemail.. I need to have seperate VM boxes for each division so that the user will be able to distinguish between the messages for each division.. Basically the only ways I can think of to make this work is to teach the users how to access each seperate VM box with a seperate VM box number and password or I could use VM contexts and have two identical VM boxes in each context but then the user will have to access VoiceMailMain differently (eg 100 for division1 VM and 101 for division2 VM) for each VM box.. Can anyone think of any easier ways? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1 user 2 VM boxes?
Senad Jordanovic wrote: Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ The single server works fine for the two divisions making and recieving calls.. Its that each extension needs two VM boxes (one for each division) thats the problem and I want to make it as simple as possible for the users to access the two seperate VM boxes from the one extension.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1 user 2 VM boxes?
Jason Williams wrote: At 11:40 29/06/2004 +0100, you wrote: Senad Jordanovic wrote: Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ The single server works fine for the two divisions making and recieving calls.. Its that each extension needs two VM boxes (one for each division) thats the problem and I want to make it as simple as possible for the users to access the two seperate VM boxes from the one extension.. Why not front the VM access number with an IVR press 1 for Division 1 press 2 for division 2 Thats not a bad idea.. Then have two identical VM boxes in seperate VM contexts with the same password.. That should work.. Now I just have to customise the email thet is sent to tell the user which divisions mailbox is the one with the message but that should be easy.. Thanks to all.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on 64bit ?
Hans-Henrik Andresen wrote: Hi, A'm about to set up a asterisk for 5000 users, and the customer had a 64bit server - can asterisk compile on that ? I will use a digium X100P for timing use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6) What else ? Is it posible to have only one server for 5000 users ? I gues that it will be 5-700 sim. users only talking sip, and IAX2 to my PSTN-Gateway. The system is suposed to scale to 15000 users. I'm ready to receive input :) /Hans-Henrik Andresen I seriously doubt 1 server will handle that type of load (unless you throw about 15 processors in it).. My advice would be to setup 1 server per 1000 users (working on appox 10 to 1 ratio of active calls) then setup IAX trunks between the servers and the PSTN gateways.. This way if a server fails then you only have 1000 users down and not the whole company.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora2 and Kernel 2.6 again!
Setting up a new system using Fedora Core 2.. Tried following the instruction below (from the mailing list archives) that worked before.. cp configs/config-for-my-kernel .config make oldconfig make include/asm make include/linux/version.h make SUBDIRS=scripts .. but now the FC2 kernel has been upgraded to 2.6.6 and these instructions no longer seem to work which means I can't get Zaptel to build.. Anyone know how to fix it? Is using kernel 2.6 just a bad idea that should be avoided for the foreseable future or will Zaptel and Asterisk be made to work with the newer kernel soon? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 2 and Kernel 2.6
Hi All, I decided to have a go at installing Asterisk on FC2 which now runs on Kernel 2.6.. Unfortunately I didn't get very far.. When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? Has anyone got Asterisk up and running on Kernel 2.6 yet? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
Joshua M. Thompson wrote: On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. Thanks for the try but its didn't work.. Got exactly the same result.. Anything else I can try? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
Joshua M. Thompson wrote: On Thu, 2004-05-20 at 12:04, WipeOut wrote: Thanks for the try but its didn't work.. Got exactly the same result.. Apparently the FC2 2.6.5 kernel has another issue, one that I didn't start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few files that are auto generated by the Makefile. The complete directions to set up your source tree are thus: cp configs/config-for-my-kernel .config make oldconfig make include/asm make include/linux/version.h make SUBDIRS=scripts A pain in the butt but at least you only have to do this once after installing a new kernel-source RPM. Well done Joshua!!.. I have no idea what all that just did but it looks like Zaptel has built.. I won't be able to test the drivers for a while with and actual card but at least i can now try build Libpri and Asterisk.. Thanks.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200
Hermann Wecke wrote: Sorry to ask this here but I believe that it is the best place to receive a feedback... I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *, and the overall impression about these phones... I am using Snom 200's and they work great.. I would guess the 100's would be just as good.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk resource consumption..
No this is not another of the What hardware do I need? posts.. :) Just wondering if anyone has calculated the memory consumprion for running asterisk.. For example, when its idle it uses U MB or RAM, uses V MB for each active Zap channel, W MB for each active SIP channel, X MB for each active IAX Channel and Y MB for each VM channel.. I often read posts where guys are throwing 1-2GB or RAM at a server and this seems extreme to me but there don't seem to be any numbers for people to use as a guide (could be placed on voip-info.org).. Unfortunately my system is to small to do any real testing of this kind but I would be happy to help somone who has a bigger more active system to get the numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk resource consumption..
Robert Lawrence wrote: Unfortunately, it is not as simple as just counting channel types. You need to factor in which compression codecs are used, as each will have a different impact on the system. Are multiple codecs used requiring transcoding, or just one? And what is Asterisk being used for? Is Asterisk monitoring the channels or is native bridging permitted? Is muisc on hold used, and if so, how many different music on hold contexts? Is Asterisk recording the channels? What about meetme conferences? Regards. All valid points.. I realise that it would not be a perfect calculation but if it was possible to say that in an average you need X MB for running Asterisk and then Y MB per active channel it would at least give people some method to calculate the memory requirements, this is similar to most software where they will say you need a certain amout of memory to run the app and then in server systems they tell you to add an amount per user.. For example, my system uses 83 MB of RAM (excluding buffers and cache) when idle and when I make a call to the echo test it uses about 20k.. This means that by these numbers a standard system for 100 concurrent users would need no more than 128MB of RAM minimum and 256MB would be plenty.. Obviously this is very simplistic and only based on a echo test but it does mean that the people who are throwing 2GB of memory at their asterisk servers are wasting a lot of money that could have been put into faster processors to handle more calls and services.. Thats really all I was talking about so that it helps people size their systems and would probably mean fewer What system do I need? questions.. Later.. WipeOut wrote: No this is not another of the What hardware do I need? posts.. :) Just wondering if anyone has calculated the memory consumprion for running asterisk.. For example, when its idle it uses U MB or RAM, uses V MB for each active Zap channel, W MB for each active SIP channel, X MB for each active IAX Channel and Y MB for each VM channel.. I often read posts where guys are throwing 1-2GB or RAM at a server and this seems extreme to me but there don't seem to be any numbers for people to use as a guide (could be placed on voip-info.org).. Unfortunately my system is to small to do any real testing of this kind but I would be happy to help somone who has a bigger more active system to get the numbers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help choosing a UK IAX provider
Steve Kennedy wrote: On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve QoS on the internet!! That will be the day.. I can see it now, all the P2P software will set their programs to run with maximum priority and then publish that they have the fastest system.. I know BT is thinking about creating QoS facilities on the DSL but its not available yet and will cost extra.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 seems healthy, but no dialtone??
Darren Nickerson wrote: Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead when I plug an analog phone in, and I don't see them go off-hook in the asterisk console when I take the phone handset off-hook. I had the same problem with my TDM400 card a little while ago.. I fixed it by using older versions of everything (in otherwords not cvs versions).. Currently using.. asterisk 0.7.2 libpri 0.5.2 zaptel 0.8.1 .. with no problems at all... Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 200 Admin Password
Hi, I have a Snom 200 that has had admin mode switched off and I have no idea when the admin password has been set to.. Does anyone know of a way to reset the phone to factory defaults?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 Admin Password
Chris Orme wrote: Hi. Did you buy the phone or get it second hand ? If second hand do you have any paperwork from the person you bought it from and did they buy it through official distribution? If you got it through distribution I would am fairly sure your vendor might be able to help ? I have a rough idea of how it would be possible but I would think you'll probably have to prove ownership as this password is how carriers lock their phones. If you got it from a carrier I imagine you might possibly have to pay them an unlock charge so you can change carriers. Or did you accidently set the admin password? Chris I bought the phone new about a year ago so its not provider locked.. I set the password to be nothing (I think) and then I set admin mode off, then when I tried to get into the admin area I couldn't, it would seem that either there is a bug that doesn't allow a blank password or it did not set it to be blank.. I will have to get hold of the distributor next week.. Later.. On Sat, 17 Apr 2004, WipeOut wrote: Hi, I have a Snom 200 that has had admin mode switched off and I have no idea when the admin password has been set to.. Does anyone know of a way to reset the phone to factory defaults?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 Admin Password
Pertti Pikkarainen wrote: There is a way. Right after reboot, and when you see the first text, hit any key and you will get a 'boot menu'. Give the phone an ip-address and define a tftp-server. The bootfile must be named snom200.bin ( e.g rename the latest snom sw ). After you have succesfully got it to download the code, the phone is also resetted to factory defaults. You will see erasing flash etc. If the download fails the phone will use the sw it has got and there will be no change in the config either. --Pertti Hmm.. That would mean I would have to setup a TFTP server which is a hassle.. :) I was hoping that there was some key combination or a reset busson hidden somewhere.. Later.. Chris Orme wrote: Hi. Did you buy the phone or get it second hand ? If second hand do you have any paperwork from the person you bought it from and did they buy it through official distribution? If you got it through distribution I would am fairly sure your vendor might be able to help ? I have a rough idea of how it would be possible but I would think you'll probably have to prove ownership as this password is how carriers lock their phones. If you got it from a carrier I imagine you might possibly have to pay them an unlock charge so you can change carriers. Or did you accidently set the admin password? Chris On Sat, 17 Apr 2004, WipeOut wrote: Hi, I have a Snom 200 that has had admin mode switched off and I have no idea when the admin password has been set to.. Does anyone know of a way to reset the phone to factory defaults?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 Admin Password
Pertti Pikkarainen wrote: There is a way. Right after reboot, and when you see the first text, hit any key and you will get a 'boot menu'. Give the phone an ip-address and define a tftp-server. The bootfile must be named snom200.bin ( e.g rename the latest snom sw ). After you have succesfully got it to download the code, the phone is also resetted to factory defaults. You will see erasing flash etc. If the download fails the phone will use the sw it has got and there will be no change in the config either. --Pertti Thanks Dude!! This worked so I have my phone back.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Hang on, we're on our way to 1.0
Olle E. Johansson wrote: We're getting closer and closer to a 1.0 release of Asterisk. In order to get there, the development is now 110% focused on solving major, critical and crash bugs. (And yes, if you follow the CVS updates, you'll see the impossible extra 10% :-) All sounds very exciting.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Redhat 9 OVER, Fidora Support, comments please.
Tony Mountifield wrote: In article [EMAIL PROTECTED], WipeOut [EMAIL PROTECTED] wrote: FC1 is basically what RHL10 would have been so compatibility is really the same as for RH9, the only issie is there appears to be an issue with the version of bison than comes with FC1 and Asterisk.. Installing the RH9 version of Bison solves the problem.. What kind of problems? I didn't notice any problems with building Asterisk on FC1 with the supplied bison, but perhaps I wasn't paying attention or didn't know what to look for. Cheers Tony For me it would bit compile with the FC1 version of Bison.. When I installed the RH9 version it worked.. Of course this issue may have been fixed in a later CVS update.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Paul Tyreman wrote: Thanks for all the replies. Can someone tell me if it is possible to put two of these X100P cards into the same machine to order to gain access to two BT landlines ? Would it also be possible for someone to outline in a bit more detail the procdue for limiting which phones have access via the card as I am new to Asterisk. What happens when someone calls the number of the line the card is on - Do all phones ring or what happens ? Is that auto attendant thing a real possiblity. What I would idealy like is this... Welcome. If you know the extention you wish to call, press * now and then dial it. Otherwise, press 1 for Family A, 2 for Family B and 3 for Family C. If the user Presses 1, Press 1 for Person A, Press 2 for Person B. etc ? Is that possible ? Thanks, Paul. It sounds like you are trying to share the PBX between multiple people.. I would suggest getting an ISDN BRI line and an AVM Fritz card (using the chan_capi driver).. This will give you two lines onto which you can get 8 MSN's (an MSN is another number coming in on the same BRI).. You can setup Asterisk to route the calls to the correct phones or group of phones based on the number that was called.. If you are in the UK there are plenty of Fritz cards around and this method will also allow you to have CallerID if you want it where the analog cards have issues with CallerID.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Paul Tyreman wrote: What I want to do is have the asterisk server sat in my house and used by my family to access the BT landline and to recieve calls made to that landline. If it is not possible to do the auto attendant thing then so be it, I will just have all phones in my house ring when a call is made on the BT line. That should be easy, right ? In addition to running the server just for my house, I want to have other memebers of my extended family link up to the server via their broadband connections so we can make free calls to each other over the internet connections. What I don't want is for other members of my family (who are not resident in my house) to be able to make calls on my BT landline, but I do want them to be able to make unlimited calls to other extentions on the asterisk server. Since I already pay monthly for broadband, I am not very keen to start paying more for an IDSN line which will only be used for this project. I don't use / need caller ID on external calls, so thats not an issue. Does that all make sence ? Thanks, Paul. What you are wanting to do is certainly doable with Asterisk.. I would suggest you read the handbook and try and understand contexts, then you should have no problems from there.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small linux distro to run * in old boxes
Victor Perez wrote: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? I have got it to install on Trustix (92MB min install) but I have moved to Fedora now for other reasons.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small linux distro to run * in old boxes
Brancaleoni Matteo wrote: I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Matteo. Are you going to be making this available or is it something yo created for inhouse use only? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restart Asterisk
Jain, Sonal wrote: Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. Thanks, No, you don't have to restart, you have to reload.. From the CLI just type reload and hit enter.. or for a command line run asterisk -rx reload.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Largescale Asterisk setup - 1000 external lines
[EMAIL PROTECTED] wrote: Hi there Does anyone know if it is possible to install a largescale asterisk cluster with up to 1000 external lines. Redundancy and loadbalancing would surely be a must for a such system, but which other things should be considered? Are you planning on using analog or IP phones for users? If IP phones you need to look at choice of codec, network bandwidth for the X000 users, how you will allocate users to servers and how users allocated to various servers will communicate with one another.. If a server fails how will you redirect the clients that were attached to that server to another server? Then there is the issue of line selection since you can't connect 1000 lines to a single server, how will this be load balanced? If you are trying to create an SSI cluster I think you have you work cut out for you because AKAIK it can't be done yet and Asterisk does not yet share registeration information between servers.. Good luck!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
James Gardiner wrote: Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Yup, this has been coming up for a while now.. Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? I have started converting my systems to it and so far I have 3 servers and my desktop running FC1.. Is the Asterisk development team got Fedora Project in mind and fully supported? FC1 is basically what RHL10 would have been so compatibility is really the same as for RH9, the only issie is there appears to be an issue with the version of bison than comes with FC1 and Asterisk.. Installing the RH9 version of Bison solves the problem.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Capacity
Adam Hart wrote: WipeOut wrote: Doesn't NuFone use SER in front of Asterisk? so using asterisk purely as the PSTN gateway.. Later Nufone offers IAX termination, SER is SIP - or am I missing something here? Sorry, I was not thinking, you are correct.. Just most termination providers (SIP obviously) seem to put SER in front or Asterisk to get the scalability.. I guess my fingers were just going faster than my brain.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STABLE 1.0 Branch CVS repository
Martin wrote: -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/03/04-10:19:04\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe Any clue's why this is happening ? Regards...Martin If you are using Fedora Linux, they ist because there seems to be an issue with Bison.. Down grade to the old RH9 Bison RPM and it should build.. If you are not using Fedora than I have no idea.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Just static on TDM400P (not even a dialtone)
For anyone who is interested.. I downgraded my system to.. asterisk-0.7.2 libpri-0.5.2 zaptel-0.8.1 +asterisk-addons ..and its all working again... Later.. WipeOut wrote: Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great.. The problem is my cordless analog phone that is connected to the TDM400P.. When I take the cordless phone off hook I don't even get a dialtone.. I only get a static thet gets louder for a second or two and then fades for a second or two and finally settles to a faint hiss.. I am not able to make a call through it and it does not ring for inbound calls.. Has anyone got any ideas what could be wrong? I am running todays CVS versions of everything except Asterisk which is a checkout of the 1.0 stable branch.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Capacity
Steven Sokol wrote: There are carriers using Asterisk to terminate thousands of lines. NuFone has a data center with 80 Asterisk servers in place. These installations require a bit more engineering than the typical PBX server, but the system does scale to extremely large systems. Steven Sokol Owner/Manager Sokol Associates, LLC Doesn't NuFone use SER in front of Asterisk? so using asterisk purely as the PSTN gateway.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Just static on TDM400P (not even a dialtone)
Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great.. The problem is my cordless analog phone that is connected to the TDM400P.. When I take the cordless phone off hook I don't even get a dialtone.. I only get a static thet gets louder for a second or two and then fades for a second or two and finally settles to a faint hiss.. I am not able to make a call through it and it does not ring for inbound calls.. Has anyone got any ideas what could be wrong? I am running todays CVS versions of everything except Asterisk which is a checkout of the 1.0 stable branch.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call forwarding / remote dial-in/out?
Angus Berry wrote: I haven't found this in any docs or faqs yet, so I'm wondering if I can achieve what I would like to do. On an Asterisk PBX with multiple PSTN lines, I'd like to call in from one PSTN line, probably via cellphone and access the PBX as if I were local to it. From here I'd like to get a dial tone and call back out. I know this isn't exactly call forwarding per se, but I'm wondering if this can be done. thanks Simply drop your inbound call to a contaxt that has access to outbound lines.. You may want to put a pin onto it so you don't get everyone using it and running up your phone bills.. You could extend on this and have a callback as well, so the yser calls in, the system then calls them back and offers them a dialtone.. Really shouldn't be hard to do, just dig around the handbook and the other Asterisk info sires and you should be able to put it together.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : MeetMe Web User Interface
Looks like a cool system.. looking forward to seeing it develop.. Later.. Areski wrote: Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has been changed. I m not providing the sources now, probably tomorrow, I really must clean the code a bit and add the mysql support (now only support postgresql I know lot of you use mysql), etc... Actually it's working really fine for me... Web UI Options already done : * control voice rights (user can talklisten or only listen) * give voice to the next user depending of the entrance order * eject user from the conference room * call out the operator of each room when an user join an empty room (Via an AGI) Web UI Options I m working on : * call out and join new participants to the conference room * ... At the end of the email, you will find the zatpel conference mode, I just implement the two masks (only listen and listen-talk): - ZT_CONF_CONFMON | ZT_CONF_LISTENER - ZT_CONF_CONF | ZT_CONF_TALKER | ZT_CONF_LISTENER Anyone needs more ?!? Well, I wanted to ask to the list if some of you are interrested to it (hope to) and if perhaps if you have some ideas concerning options that we can add or whatever to make this application better. And if anyone need special custom/features, feel free also to contact me off list. I welcome feedback, Cheers, Areski #define ZT_CONF_MODE_MASK 0xff /* mask for modes (255) */ #define ZT_CONF_NORMAL 0 /* normal mode */ #define ZT_CONF_MONITOR 1 /* monitor mode (rx of other chan) */ #define ZT_CONF_MONITORTX 2 /* monitor mode (tx of other chan) */ #define ZT_CONF_MONITORBOTH 3 /* monitor mode (rx tx of other chan) */ #define ZT_CONF_CONF 4 /* conference mode */ #define ZT_CONF_CONFANN 5 /* conference announce mode */ #define ZT_CONF_CONFMON 6 /* conference monitor mode */ #define ZT_CONF_CONFANNMON 7/* conference announce/monitor mode */ #define ZT_CONF_REALANDPSEUDO 8 /* real and pseudo port both on conf */ #define ZT_CONF_DIGITALMON 9/* Do not decode or interpret */ #define ZT_CONF_FLAG_MASK 0xff00/* mask for flags (65280) */ #define ZT_CONF_LISTENER 0x100 /* is a listener on the conference (256)*/ #define ZT_CONF_TALKER 0x200/* is a talker on the conference (512)*/ #define ZT_CONF_PSEUDO_LISTENER 0x400 /* pseudo is a listener on the conference (1024)*/ #define ZT_CONF_PSEUDO_TALKER 0x800 /* pseudo is a talker on the conference (2048) */ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup not detected on X100P
Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Funny you should say that.. I have just updated my system to the latest CVS of zaptel, libpri asterisk and asterisk-addons and now it is screwing up.. If a call comes in on the X100P the sip phones (3 of them) and one analog cordless phone (connected to a single port TDM400P) ring.. When I answer one of them the others continue to ring.. I have to go to each one and answer them to stop them ringing but I cannot answer the call.. The line is then tied up and I have to kill the server to get it cleared.. If I try and make a call I cannot hear any audio on the sip phones and when I hang up either side the line stays connected.. I thing the current CVS is broken.. What the latest stable version? 0.7.2 or 1.0 Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple IAX register lines?
Hi, Having a small problem here and wondering if anyone else has seen it.. My Asterisk box is behind NAT so I need to register with the external IAX Asterisk boxes for calls to be received.. Up till yesterday I only needed to register with a single external IAX server and all was working fine.. Now I need to register with a second external IAX server.. So I now have two register lines in my IAX.conf.. The problem is that Asterisk only uses the first one and is ignoring the second.. If I comment out the first one then asterisk will happily use the second one but it does not seem to be happy using both at the same time.. Is anyone else having this problem? is it a bug? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
[EMAIL PROTECTED] wrote: Works like a charm for me. I have both VoicePulse and NuPhone registered in IAX. Depending upon the phone nr dialed, I send a call via NP or VP. And yes, my [*] box is behind a NAT. Include the relevant lines of your iax.conf so we can take a look. Cheers, Willy There is nothing to it really register lines are pretty simple.. register = user1:[EMAIL PROTECTED] register = user2:[EMAIL PROTECTED] From the cli iax2 show registry only shows the first entry.. These are for inbound services not outbound, I didn't think it was nesesary to register for outbound calls because the call is being initiated from inside.. Later.. - Original Message Follows - Hi, Having a small problem here and wondering if anyone else has seen it.. My Asterisk box is behind NAT so I need to register with the external IAX Asterisk boxes for calls to be received.. Up till yesterday I only needed to register with a single external IAX server and all was working fine.. Now I need to register with a second external IAX server.. So I now have two register lines in my IAX.conf.. The problem is that Asterisk only uses the first one and is ignoring the second.. If I comment out the first one then asterisk will happily use the second one but it does not seem to be happy using both at the same time.. Is anyone else having this problem? is it a bug? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the registration
process. Simply set up a type=user (for inbound iax calls) with some
appropriate security parameters (host, secret, context). The associated
extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2 servers,
or if any site is using dynamic IP addresses, then your stuck with
resolving the original registration problem. (Does a iax2 debug show
anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
Later..
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Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote: Thats the problem I have a dynamic IP on my side which is why I need the register line in the iax.conf.. iax2 debug shows that it is registering the first line but not the second.. I assume you've tried the easy stuff... separate the two statements with some additional cf-lf, add a useless/redundant statement in between the two registrations, move one of them to the bottom of the config file, database show to ensure entries are accurate, etc, etc. Yup done all that.. makes no difference.. I have just built from the latest CVS as well and its still not registering to the two servers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Hi,
I have just solved it, I deleted the iax.conf and created a new one
(exactly the same) and now its working.. I guess there was a gremlin in
the text file somewhere the was twisting things up..
Thanks to everyone for you help and suggestions..
later..
Ed Rubright wrote:
In you iax.conf file, are you using type=friend? I seem to remember a
discussion about problems if using type=friend instead of type=user for
multiple registered servers with the same userid. I may have the details
messed up on this though!
I have the same setup your describing and it works great!
In the iax.conf file I have broken the inbound and outbound into 2 separate
stanzas, i.e. -
; for inbound from Nufone
[NuFone]
Type=user
; for outbound to Nufone
[NuFone-peer]
Type=peer
Hope this helps.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, March 26, 2004 5:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED] register =
user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the
registration process. Simply set up a type=user (for inbound iax calls)
with some appropriate security parameters (host, secret, context). The
associated extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2
servers, or if any site is using dynamic IP addresses, then your stuck
with resolving the original registration problem. (Does a iax2 debug
show anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
Later..
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Re: [Asterisk-Users] Asterisk
simprix wrote: What kind of specs do I need for a asterisk box that will have a pri for pstn and about 65 sip phones I was thinking a Xeon 3.05 What length is a piece of string when you cut it? I was thinking 2.374 m Sorry about the sarcastic answer but if you look through the mailing list archives you will see that this question gets asked all the time with and the person asking never gives enough information.. Things like are you using IP phones or analog phones? if IP phones what codecs do you plan to use? how many concurrent calls to you expect? etc.. Its difficult to give someone an estimation of load with no information.. Maybe this will help you.. http://www.voip-info.org/wiki-Asterisk+hardware+recommendations Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Peer Oliver schmidt wrote: Brian Capouch wrote: It might be helpful for us all if the author could let us know more about the environment in which this application was built. . I'm getting all kinds of errors when I try to run it, and I suspect that either my Postgres or PHP installations are incompatible with yours. Another thing I had to do was changing the defines.php file to reflect my environment. After that, things went smooth. On my server the links dont even work in the menu on the left.. Not sure what is going on with the code and dont have the time to look right now.. I will just have to wait for the next version.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Peer Oliver schmidt wrote: Wipeout wrote: Another thing I had to do was changing the defines.php file to reflect my environment. After that, things went smooth. On my server the links dont even work in the menu on the left.. Not sure what is going on with the code and dont have the time to look right now.. I will just have to wait for the next version.. :) The links don't work because of a wrong setting in defines.php. Just put the path on your webserver for the files and things should work. On the other hand, waiting for something new is another option ;-) rgds pos Just double checked, my paths are correct in defines.php.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones can talk to asterisk but not each other through it
Tony Mountifield wrote: I posted this a week or two ago but no replies, so trying again... Summary: Two phones in different locations, each behind NAT, can both talk to an Asterisk server on the net, for the demo or for voicemail, but can't maintain a call to each other via that asterisk. Original post with details: I have a problem with an installation of asterisk on my colo server. I have a Grandstream BT102 behind a Linux NAT firewall, and my colleague also has one behind his. My connection is ADSL with 512k down and 256k up. My colleague's is Cable with 600k down and I don't know whether it's 128k or 256k up. I have the phones set up in sip.conf with nat=yes, qualify=yes and canreinvite=no. Each phone can successfully connect with Asterisk and dial the Asterisk Demo, leave and pick up voicemail, etc. However, if one phone tries to dial the other, once the called phone is answered, the audio starts off very stuttery and broken, and after a few seconds dies completely and the call gets dropped. In the asterisk log there are many entries for that time saying: Recv error: Resource temporarily unavailable. I am using the zaprtc timer module on the asterisk server, but in any case I understood that was only required for MeetMe or MOH. The server system is a Duron XP 1800, with 512MB RAM, running Fedora Core 1 with updates, and a standard 2.4.22 kernel that was recompiled only to make the RTC a module instead of compiled in (so I could rmmod it and then load zaprtc instead, which works fine). Can anyone suggest what things I should check or change? Cheers Tony Have you setup any port forwarding on the NAT boxes? If not try it, it may help.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
Daniel Bichara wrote: WipeOut wrote: Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. Sorry but I disagree. I am using SPEEX and voice quality is much better than GSM and it consumes less bandwidth. Using Speex and Linux is just a make; make install. Although, we MUST encourage OpenSource initiatives or we will pay Licenses forever. Take a look at G.723 or G.729, for example. ITU released this protocols many years ago and we still have to pay royalties. About this kind of license there is another thread talking about... Daniel Agreed, speex is fine when going * to *.. UA to * over speex is usually a nightmare, and the number of UA's with speex support are very few.. Maybe if there was more support for it in the industry it would be worth using.. We will just have to wait and see what the future holds.. Anyway personal favorite is iLBC but there is not much support in the UA's for that either.. :( So for now its GSM or G.711 from UA to *.. What a choice!! Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)
Brian Capouch wrote: Olle E. Johansson wrote: Do *not* send out personal replies on the list. Yes! Yes!! Yes!!! Let's change the way the list software works so people won't get hammered by replying and rid this list of that pox once and for all. B. The biggest problem with having replies go back to the original poster instead of the list is that on one else will be able to learn from the answers to question that was asked, and the mailing list archives will become useless as a source for information.. If that were the case a web forum would then become a better solution as a pure support resource .. and I hate web forums.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP and *
Alessio Focardi wrote: Quick hint: do I need cgi or cli version of php to interact with asterisk agi ? I'm using cgi now, with strange results tnx ! You need the CLI binary for PHP to work with AGI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with MySQL on Redhat 9
Umar Sear wrote: Hi I really hope somebody can help me out. I have an asterisk installation working on a Redhat 9 system. I now want to add the MySQL functionally to it. However when I make the necessary changes, (downloading the add-ons, and changing the Make file) the make fails. I have looked into this and I think I know what the problem is. Basically I only have MySQL binaries installed. Can anyone advice me what packages I need to install to get this going. Help will be greatly appreciated. Umar. You need to install the mysql and mysql-devel packages and any dependencies.. if you want the server to run on the same PC then you need to install mysql-server as well.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone?
I thought thats what http://www.iaxtel.org was all about.. :) Daniel Bichara wrote: Hi All, We know everyone can offer services. May we build a interconnected * network all over the world to offer best conditions each other? We can set a service level agreement and try ;-) Any one? Daniel [EMAIL PROTECTED] wrote: Since everyone is offering their services then: USA - £0.016 (~ 2.9c) UK - £0.016 (~ 2.9c) Europe - £0.02 (~ 3.6c) UK 0800 - FREE SIP / IAX termination. auto-provisioning, web-based billing, call history, on-line top-up, credit-card payments. Not US-based though :-( Tan www.voiptalk.org www.iaxtalk.co.uk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer Sent: 17 March 2004 19:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NuFone? Doug Harris wrote: Hi, Seems like there arn't any alternative to NuFone either ? Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached. Doug If you want SIP/IAX termination from someone other than NuFone for the same price, you can contact me. We can offer that. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk MySQL
Joao Carlos Moura wrote: I need to develop an web interface to include clients automatically in Asterisk. So, to make this possible I need that all my peers and exten being at a database (Mysql). Where do I find doc´s regarded for it? Thank you very much, J Moura I think MySQL friends is probably what you are after.. take a look at www.voip-info.org Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
