Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
2008/10/23 Kristian Kielhofner [EMAIL PROTECTED] Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. CISCO PIX's SIP fixup stuff breaks authentication from a SIP device if the SIP device is using an IP address for the proxy and not a DNS name. This is because the PIX rewrites the proxy's IP address where-ever it is seen. And that includes inside the authentication challenge line. (The PIX appears to do a literal search-and-replace in the SIP headers). Which means the authentication fails. We hit this twice with customers. Unfortunately long enough apart that I had to debug it all over again because I forgot about it... The workaround is to use a DNS name to address the proxy. So its definitely not just Sonicwall. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on ISDN and Asterisk in the UK
Gordon Henderson wrote: On Fri, 24 Oct 2008, Alan Lord wrote: snip / I used to have an ISDN-2 line into my home office. BT wrote to me about 2 years ago and said they were discontinuing the service. They converted my dual channel BRI back into a single POTS. Sure it was ISDN2e and not Home or Business Highway? They killed off an the HH and BG lines some time back and converted them back to POTS. I've no idea why - I'd cancelled my HH line some time before the cut-off date. Yes, I think it was called Home Highway (but it *is* just a BRI at the end of the day). It had a USB port on the NTE which I never really grokked. Not being a Windows user and all. I built a little Asterisk server, stuck an X100p in it for backup calls should my broadband go down (on a separate POTS line) and got two non-geo 0844 IAX trunks for free instead. Who lost out there then? Well, quite. BT have their good points, but also their stupidly bad points too. They phone me up once a month at present and ask me why I'm not placing any outgoing calls with them. When I try to tell them why, (because I run my own phone company!) because my reply is not in the script, they just hang up on me. LOL - That's a good one. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank loop current adjustment
2008/10/23 Udo Schacht-Wiegand [EMAIL PROTECTED] For a door opener on an Astribank FXS port we need a loop current of 24.5mA . It does not function with the Astribank now, the dialtone becomes quiet immediately after pressing the button on that device. I've seen a limit of 23mA in the zaptel source. Is it possible to change the loop current of the Astribank somehow? You need to ask Xorcom that question. They have the ability to adjust any registers in the FXS port chip so I expect that they can do that. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fresh installed box
Thanks Matt, would you please tell me in details about the following 1- the Linux mail configuration steps to enable it to send voicemail to email. 2- the steps to use T.38 and pass thru...or Fax detection...and fax to email. 3- for the live monitoring.i wanna a software to monitor and to make spying on the calls, etc... if you will send me helpful documents , your help will be appreciated Thanks, Torintino From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Oct 2008 21:45:22 -0400 Subject: Re: [asterisk-users] Fresh installed box http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on ISDN and Asterisk in the UK
On Sat, 25 Oct 2008, Alan Lord wrote: Gordon Henderson wrote: On Fri, 24 Oct 2008, Alan Lord wrote: snip / I used to have an ISDN-2 line into my home office. BT wrote to me about 2 years ago and said they were discontinuing the service. They converted my dual channel BRI back into a single POTS. Sure it was ISDN2e and not Home or Business Highway? They killed off an the HH and BG lines some time back and converted them back to POTS. I've no idea why - I'd cancelled my HH line some time before the cut-off date. Yes, I think it was called Home Highway (but it *is* just a BRI at the end of the day). It had a USB port on the NTE which I never really grokked. Not being a Windows user and all. Yup. Home Highway. It was good at the time - I had it, so I could have what was effectively a 64K leased line Internet connection and still make calls. ADSL Broadband more or less killed it. You didn't quite get all the features of ISDN2e that I recall, however I don't ever recall ever wanting to use those features... And it was home powered rather than line powered, but that's progress for you... I built a little Asterisk server, stuck an X100p in it for backup calls should my broadband go down (on a separate POTS line) and got two non-geo 0844 IAX trunks for free instead. Who lost out there then? Well, quite. BT have their good points, but also their stupidly bad points too. They phone me up once a month at present and ask me why I'm not placing any outgoing calls with them. When I try to tell them why, (because I run my own phone company!) because my reply is not in the script, they just hang up on me. LOL - That's a good one. I know - they always tell me that the call may be being recorded too. I put them out of their misery by teling the call definately is being recorded - by me :) BT - Great bits of technology, forward thinking network features, etc. (Eg. TXT to speech as mentioned in another thread here), however they're driven by corp-rat decision makers being poked by greedy share-holders )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank loop current adjustment
On Thu, Oct 23, 2008 at 02:50:05PM +0200, Udo Schacht-Wiegand wrote: For a door opener on an Astribank FXS port we need a loop current of 24.5mA . It does not function with the Astribank now, the dialtone becomes quiet immediately after pressing the button on that device. I've seen a limit of 23mA in the zaptel source. Is it possible to change the loop current of the Astribank somehow? A phone that is on-hook is supposed to be an open circuit and ideally not draw any current. In the real world this doesn't really happen. The Astribank detects this by setting two limits. Anything above the high limit is considered closed circuit (off-hook). Anything below the lower limit is considered open circuit (on-hook). The values of those limits are 6.72mA (low) and 10.16mA (high). Thus 24.5mA is well into what is considered an open circuit. If you try to raise those limits, you may encounter problems with detecting a phone being picked up and such. As Steve mentioned, tweaking this can be done by setting registers. However those tweaks tend to interact with other things in strange ways. Therefore I would not provide here the specific details without fully understanding the situation. Generally for such matters contact [EMAIL PROTECTED] . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on ISDN and Asterisk in the UK
On 24 Oct 2008, at 17:00, Phil Knighton wrote: Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've even offered us free installation of the lines (as opposed to a £560 charge for moving the analog lines!) What hardware would you recommend in the Asterisk box? I don't mind admitting I'm a newb and a lot of the info I've found is over my head. I've been looking at a TE410P - would this achieve what I want which is to connect the 3 ISDN2 connections, giving me six voice channels? Assuming the TE410P is what I'm looking for (or an equivalent - suggestions?) what are the basic points for what I would need to change in my current config? Any help or suggestions would be gratefully appreciated :-) Cheers Phil Couple of things to look out for : 1) FAX! If you currently have a fax on any of the 4 analog lines, then moving to ISDN will require you to do a major dance to get it working with any degree of reliability. (same goes for dialup modems, creditcard processing machines, alarm systems, sky boxes etc) 2) There really isn't any competition in the ISDN BRI market in the UK - once down that road you are tied to BT. If you do want to go to ISDN, think seriously of getting a PRI (30 channels) with only a few channels 'lit'. The normal minimum is 8 out of the 30, but I once persuaded NTL (as was) to put up a 6 line PRI. All the telcos have PRI offerings, and the cards for asterisk are cheaper than the equivalent BRI card. In your situation I'd be taking 2 analog lines and an ADSL, use the ADSL to make voip calls through a good VoIP provider and the 2 analog lines for emergencies, faxes and failover. Tim.___ -- 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] The skype channel...
Hello everyone! Perhaps I missed something: But where can one download the beta-version of the new asterisk skype channel? Can it work with 1.6.0-beta9? I tried to browse the digium downloads, but it's dificult, if you're blind and only have a text-based (almost no javscript) browser. Thanks for any good hints and pointers! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
On Friday 24 October 2008 11:49:15 am Wilton Helm wrote: 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. fax is a legally accepted form of document transport in many states. signing and returning mortgage/lease/title/contract papers during the sale of a house for example. theoretically fax has two legally defined and assignable endpoints--they can be identified with a person or an organization. though i do wish fax would go away in favor of real (affordable) electronic signatures or universally accepted gpg -- we all get one with our social sec. numbers or something, but this is nation-dependent. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] The skype channel...
You had to sign up with a form and Digium was going to get back to you. Don't know if they got back to people yet..but I didn't hear from them yet. I don't have the form url anymore. Regards, Dean Collins Cognation Inc [EMAIL PROTECTED] +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julien Claassen Sent: Saturday, 25 October 2008 9:56 AM To: asterisk users mailinglist Subject: [asterisk-users] The skype channel... Hello everyone! Perhaps I missed something: But where can one download the beta-version of the new asterisk skype channel? Can it work with 1.6.0-beta9? I tried to browse the digium downloads, but it's dificult, if you're blind and only have a text-based (almost no javscript) browser. Thanks for any good hints and pointers! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The skype channel...
Dean Collins wrote: You had to sign up with a form and Digium was going to get back to you. Don't know if they got back to people yet..but I didn't hear from them yet. I don't have the form url anymore. Regards, Dean Collins Judging by their emails, it looks like at least the first wave of beta testing has begun. ___ -- 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] The skype channel...
Hi! So no way to get in anymore? that's too bad. It would have been the first real skype accessibility for blind people working like myself on linux. If anyone does remember or can retrieve the URL for the signup form, please tell me anyway, perhaps it's still possible. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The skype channel...
On Sat, Oct 25, 2008 at 4:58 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! So no way to get in anymore? that's too bad. It would have been the first It's in the first wave. Here's what the initial mail from Steve Sokol said: The Skype For Asterisk beta will start with a small group and expand in a controlled manner to allow close monitoring of feedback and optimal operation between Skype and Asterisk. An email notice informing those selected for the initial phase will be sent out some time late this week. If you are selected you will receive a beta license agreement and a mutual nondisclosure agreement, both of which must be signed before you can download the beta code. If you are not selected for the initial round please keep in mind that our goal is to expand the beta program as soon as is practical. We will keep your application and when we get to the second phase of the beta you will be the first to know. That note was sent just after Astricon. If you applied you should have received it. /r ___ -- 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] The skype channel...
Julien Claassen wrote: Hi! So no way to get in anymore? that's too bad. It would have been the first real skype accessibility for blind people working like myself on linux. If anyone does remember or can retrieve the URL for the signup form, please tell me anyway, perhaps it's still possible. Kindest regards Julien Send em your details on the form on their website, there is more than one round of beta testing. ___ -- 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] Returning to Voicemail after returning call
No, it is not possible. I submitted a bug report[1], because it has been bothering me too. [1]http://bugs.digium.com/view.php?id=13781 On Thu, Oct 23, 2008 at 4:36 PM, Mark Wiater [EMAIL PROTECTED] wrote: Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cheapest 4 port FXO
I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- 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] someone to test gtalk with me?
Hello everyone! Would someone be willing to test googletalk with me? Please reply in private, I'll get you my account and we can exchange anything else, that's necessary. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fresh installed box
Hi Torintino, 1. Login to FreePBX, Go to extensions, Select the extension you want to configure, Scroll down to the bottom under the voicemail setup section, and check the Attach to Email checkbox and then save the extension and reload freepbx. Now your emails will be sent including the voicemail. Note that mail has to be setup on the box for it to work (ssmtp or local mta). 2. Here are some tutorials - http://www.voip-info.org/wiki-Asterisk+fax, http://www.voip-info.org/wiki/view/T.38 http://nerdvittles.com/index.php?p=88 http://asterfax.sourceforge.net/ 3. Ah, I'm not positive on what would work for this - sounds like some modifications to FOP may be in need. Maybe someone else on the list has ideas. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Torintino T Sent: Saturday, October 25, 2008 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box Thanks Matt, would you please tell me in details about the following 1- the Linux mail configuration steps to enable it to send voicemail to email. 2- the steps to use T.38 and pass thru...or Fax detection...and fax to email. 3- for the live monitoring.i wanna a software to monitor and to make spying on the calls, etc... if you will send me helpful documents , your help will be appreciated Thanks, Torintino _ From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Oct 2008 21:45:22 -0400 Subject: Re: [asterisk-users] Fresh installed box http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-q ueuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista Learn more! http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gtalk dialstring?
Hi everyone! I couldn't find anything expressive about gtalk dialstrings. It doesn't seem to work. I'm not sure why, so I'll start at the easiest point. The syntax I found was: gtalk/my_account_name/[EMAIL PROTECTED] Is this correct? And does any of you googletalkers know, if a simple google-mail account is enough to use the talking bit, or do I have to register in the googletalk software again? Oh btw. Here's my error: bach [Oct 25 21:18:11] ERROR[28847]: chan_gtalk.c:908 gtalk_alloc: no gtalk capable clients to talk to. [Oct 25 21:18:11] NOTICE[28847]: channel.c:3243 __ast_request_and_dial: Unable to request channel gtalk/gtalk_account/[EMAIL PROTECTED] Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk dialstring?
Hi Julien, bach [Oct 25 21:18:11] ERROR[28847]: chan_gtalk.c:908 gtalk_alloc: no gtalk capable clients to talk to. [Oct 25 21:18:11] NOTICE[28847]: channel.c:3243 __ast_request_and_dial: Unable to request channel gtalk/gtalk_account/[EMAIL PROTECTED] The syntax is correct. Make sure that you have the [gtalk_account] section inside your jabber.conf file, you can also check the connection to the GoogleTalk XMPP server by issuing these commands : jabber show connected jabber show buddies (in Asterisk 1.6) Cheers, Philippe ___ -- 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] gtalk dialstring?
Hi! OK, this seems to work. But still I can't find anything to talk to for a test. I don't know, if the firewall might be in the way. Isn't there some echo-test service for google? And about the buddies: I found usernames in two places 1. in jabber.conf [gtalk_account] ... [EMAIL PROTECTED] 2. gtalk.conf [guest] disallow=all allow=ulaw context=google-in [EMAIL PROTECTED] Do I need both? How to add more buddies? Just go on with: [EMAIL PROTECTED] and [EMAIL PROTECTED] or do I have to use something like buddy2 and suername2? google-in is the context of incoming calls in my extensions.conf, is is correct to be there as a context name, or should there be some other context? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cheapest 4 port FXO
X100P. Joseph L. Casale wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cheapest 4 port FXO
X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. Thanks! jlc ___ -- 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] Agents log in afterhours
On Fri, Oct 24, 2008 at 9:33 PM, Ing. Jorge S Alanís Garza [EMAIL PROTECTED] wrote: Hi all, I received a report of a client which stated that two of its agents are logging in to the queues when they actually aren't there working. They appeared to be logged on all night. They thought they weren't logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours? The question is actually - what impact does it have. Agents could login in working time and just forget to log out. So any deny would be ineffective. We have DID routing determined by free member count in queues + working hours. Some configurations allow calls to go to queue within working hours if there are no agents, and some don't allow to accept calls in after hours. So, in order to not have callers wait in queue within afterhours we have several methods used together: 1) If agent don't answer a call, he gets either paused or logged off. Paused agents don't count as free, but they are still around, so routing might send call to IVR first to welcome caller and give agents some time. 2) Within after hours all agents are logged out every 15 minutes. So, they are allowed to work after official working hours, but they just have to relogin every 15 minutes. Realtime queue members in MySQL and cron script makes this quite straightforward :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Strange ring tone: Long-Short-Short
I'm using Linksys SPA3102 adapter and have a strange ring tone: Long-Short-Short or Long-Long-Short-Short Does anybody know which setting adjust this ring tone on SPA3102 Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Warning messages
Hello guys, I am having some problems when user Queue app, the problem is that the celler do not hear the queue messages like queue-periodic-announce, etc and I am having this two warnign messanges when that happends, what does this messages means ? First One: Started music on hold, class 'default', on DAHDI/1-1 [Oct 25 15:23:18] WARNING[10216]: channel.c:1893 ast_waitfordigit_full: Unexpected control subclass '2' Second One: -- DAHDI/1-1 Playing 'queue-periodic-announce' (language 'en') [Oct 25 15:23:38] WARNING[10216]: file.c:1204 waitstream_core: Unexpected control subclass '2' thank you cesar -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hammering imap vmail storage
I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b. signature.asc Description: This is a digitally signed message part ___ -- 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] bug in Asterisk 1.4.22?
Hello is my idea or this is a bug? The thing is that I have in my asterisk.conf this: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /usr/local/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk where the dir of agi-bin is in /usr/local/share/asterisk/agi-bin and inside agi-bin directory I have a file called a2billing.php and in my extesions.conf i have: [a2billing] exten = 1,1,answer exten = 1,2,Wait,2 exten = 1,3,DeadAgi,a2billing.php exten = 1,4,Wait,2 exten = 1,5,Hangup and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' So, i change the file a2billing.php to another place and I change this new place in asterisk.conf: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /new/place/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk I reload the asterisk server and the asterisk log still say me the same place before: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' Why is that? Any suggest? Thanks for all, Abel ___ -- 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
