Re: [asterisk-users] Re: What's the story with X10*P FXO cards?
On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote: snip Thanks for the info. This little experiment is getting expensive ;-) LOL! I know that feeling... I actually thought I would save money with VOIP, what a joke! Actually I am happy with my setup, but spent an intial $75(us) thinking the HT-488 would provide an FXO for my PSTN connection, only to find it doesn't work well enough to actually use for that. I then spent an additional $200(us) for the Wellgate 3701a (1FXS+1FXO) which, although horribly documented is a HUGE improvement over the Grandstream garbage (HT-488). So yeah, it gets expensive quickly. It's annoying that so many of the products are marginal or straight up garbage, but I guess we are still in the infancy of the standards based VOIP business, and many of the products are in search of a firmware update, or a decent manual. Oh well, More work for the stubborn techo-maniacs who refuse to give up(like me). ___ --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] Freeware sip/iax client windows mobile
What are the min cpu requirements for ppciax? Has any one tried ppciax with Cingular 8125? On 7/9/06, Administrator TOOTAI [EMAIL PROTECTED] wrote: Attilla De Groot wrote: Hi all, I have two pda's and I want to be able to make calls, but I need a client for this. The only problem is Windows Mobile 5.0, I can't find a freeware client for this, the only one is Sjphone. But this one is still beta for windows mobile and it just doesn't work good. Does anyone have an alternative ? I'm using ppciax -- Daniel ___ --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
Re: [asterisk-users] intel vs amd motherboards
ooops, sorry, you right, forgot to mention it... It was to be compared with AMD 64. Olivier C F a écrit : Olivier can you please do a cat /proc/cpuinfo and post it here? I think you have a 64 bit cpu. On 7/9/06, olivier.taylor [EMAIL PROTECTED] wrote: Fyi, Double Intel Xeon 3Ghz performance below g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 410 2914 ulaw - 2 - 1 2 2 1 410 2914 alaw - 2 1 - 2 2 1 410 2914 g726 - 2 2 2 - 2 1 410 2914 adpcm - 2 2 2 2 - 1 410 2914 slin - 1 1 1 1 1 - 3 9 2813 lpc10 - 3 3 3 3 3 2 -11 3015 g729 - 3 3 3 3 3 2 5 - 3015 speex - 3 3 3 3 3 2 511 -15 ilbc - 3 3 3 3 3 2 511 30 - Olivier Tzafrir Cohen a écrit : On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. ___ --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 ___ --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] CallerID
On 7/10/06, Ryder Brook [EMAIL PROTECTED] wrote: and learning a lot and the stupid mistake was that the telephone that I was calling from has caller id blocked. Well, the only satisfaction is that I You should always have a way to test with a call that you know is working, such as a cell phone if you have one or an associate or firend. More and more people block CID these days, and as a counter measure, I've noticed more people filtering their calls with a message about how they won't accept calls with CID blocked! ___ --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] IVR DTMF
Dear I want to make billing recharge through receiving digits from IVR through dtmf and store it on a text file , How can I do that ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Re: What's the story with X10*P FXO cards?
Martin Joseph wrote: On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote: snip Thanks for the info. This little experiment is getting expensive ;-) LOL! I know that feeling... I actually thought I would save money with VOIP, what a joke! Actually I am happy with my setup, but spent an intial $75(us) thinking the HT-488 would provide an FXO for my PSTN connection, only to find it doesn't work well enough to actually use for that. I then spent an additional $200(us) for the Wellgate 3701a (1FXS+1FXO) which, although horribly documented is a HUGE improvement over the Grandstream garbage (HT-488). So yeah, it gets expensive quickly. It's annoying that so many of the products are marginal or straight up garbage, but I guess we are still in the infancy of the standards based VOIP business, and many of the products are in search of a firmware update, or a decent manual. Oh well, More work for the stubborn techo-maniacs who refuse to give up(like me). I've gone through the same basic devices over the last three years, but have also tested the Mediatrix 1204 box as well. I've got four analog pstn lines here coming from two different central offices, both of which are on relatively long loops with somewhat unusual echo characteristics. I have also gone through the spa3000, ht488, x100p's, multiple h/w versions of the TDM04b, etc. Also have a TDM2400 here for testing, but testing has been held up due to delays in being able to run current trunk code on the test machine. To date, the best devices from a quality of audio perspective are the Mediatrix 1204 and the Sangoma A200D. The last time I worked with the 1204 was about two years ago and the box had several compatibility and security issues that hopefully have been resolved since then. Its retail price is higher then the A200D (in the US). The A200D has been in use now for several months and has provided excellent audio (etc), supports modem use (fax and POS), and just hasn't failed in any way. Rumor has it that Digium will be announcing some new cards in the near future that should also help address the space for small numbers of analog pstn lines. Unknown as to what the current status happens to be. ___ --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] Urgent Upgrade
Dear I am in need urgently to upgrade my [EMAIL PROTECTED] from 2.6 to trixbox or 2.8 how can I do that . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Re: What's the story with X10*P FXO cards?
On Jul 9, 2006, at 11:44 PM, Rich Adamson wrote: Martin Joseph wrote: On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote: snip Thanks for the info. This little experiment is getting expensive ;-) snip Rumor has it that Digium will be announcing some new cards in the near future that should also help address the space for small numbers of analog pstn lines. Unknown as to what the current status happens to be. PCI card aren't an option for me, as I am using OSX as my Asterisk platform, so that doesn't work. I am still keeping my eyes open for additional devices that might work as an FXO or 2, and have the desired characteristics. The Echo can in the Wellgate 3701a is clearly much better then the Grandstream one, but it still takes some number of seconds to train at the beginning of calls, which is kind of funky. Thanks for the info Rich. Marty ___ --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] intel vs amd motherboards
On Sun, Jul 09, 2006 at 05:56:49PM -0400, C F wrote: Thanks for that Tzafrir. Why does it ignore the secend CPU? 'show translations' is done by a loop that for each pair of codecs meassures the time it takes to convert a relatively short ammount of data between the two. Thus each conversion is done by a single CPU. I'm not saying 'show translations' is anywhere near useless. It is a standard benchmark that comes with Asterisk and useful as such. Benchmark can be handy when used right. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] spa941 call pickup?
I've been using *8# on my 7960's to pickup ringing phones in the office. Anyone been able to do call pickup from a spa941? ___ --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] Urgent Upgrade
On Mon, Jul 10, 2006 at 09:47:34AM +0300, Khaled Chehab wrote: I am in need urgently Hire someone to do that if it is that urgent? to upgrade my [EMAIL PROTECTED] from 2.6 to trixbox or 2.8 how can I do that . Ask in the [EMAIL PROTECTED] / trixbox mailing list(s)? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] asterisk to mobile phone
Have a look at cyber-telecom.net. CT-GSM-1000 seems to be one of the cheapest GSM Gateway that you can buy right now. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Tuesday, June 27, 2006 11:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk to mobile phone I use an Ateus VoiceBlue which allows you to do this (never tried it though) which is a SIP device and you write your dialplan to send calls to the SIP device just like ringing an extension in Asterisk. It works fine but it tends to drop calls under load so I have an AGI that determines the load and if it goes beyond a certain threshold it relays the calls out the PSTN. By load, I mean as much as four calls hitting it simultaneously, if it only gets onsie-twsie calls it works fine. -Original Message- From: Lito Lampitoc [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 27, 2006 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk to mobile phone what brand of gsm gateway do you think works well with asterisk? On 6/27/06, Colin Anderson [EMAIL PROTECTED] wrote: A GSM gateway will allow you to specify a ruleset so a channel on the gateway is always locked to a particular mobile number, then you just send the call from Asterisk to the gateway and it will do the hunt for you. -Original Message- From: Lito Lampitoc [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June 27, 2006 7:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk to mobile phone Is it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup? Thanks in advance. Lito ___ --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
Re: [asterisk-users] spa941 call pickup?
Rich Adamson wrote: I've been using *8# on my 7960's to pickup ringing phones in the office. Anyone been able to do call pickup from a spa941? Disregard; dumb mistake on my part. Forgot to add pickup to the sip.conf definitions for the extension. ___ --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] setting up an email to fax with asterisk
One of my customers decided to allow me to make a test system for a fax server. So far I have searched the wiki and came up with Hylafax(standalone or with IAX) and astfax (integration with asterisk). Scenario: Customer has windows machines (500+) and we want to try a fax server in the email-to-fax fax-to-email mode with minimum intrusion in the windows machines. astfax looks promising but it uses openoffice libraries to do conversion from .doc or other formats to tiff. The thing with this is that OO sometimes lacks the reliability to do a true conversion on MS Office formats like fonts or spacing or tabs. So it will look good on MS Word for example, but crap after OO conversion. I have no intention to start a war on this, but those who use MS Office and OO will know that true font/spacing/etc conversion is far from perfect, specially when mixing different MS Office version (95,2000,XP) AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. Any suggestion for this kind of setup? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] setting up an email to fax with asterisk
So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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
SV: [asterisk-users] setting up an email to fax with asterisk
Hello If you look at hylafax.org, you can find several windows clients for Hylafax. http://www.hylafax.org/content/Desktop_Client_Software Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Erick Perez Sendt: 10. juli 2006 09:52 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] setting up an email to fax with asterisk So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07-07-2006 ___ --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] setting up an email to fax with asterisk
We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] FXS: No ringtone
Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] AGI tutorials
Anybody who knows a good source of AGI tutorials on the net? plz share-- RegardsRizwan HishamSoftware Engineer ___ --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] setting up an email to fax with asterisk
We used? what are you doing different now? On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
A different job PaulH On Mon, 2006-07-10 at 03:34 -0500, Erick Perez wrote: We used? what are you doing different now? On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 ___ --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] CDR calls started via AstManProxy
Hi, Does Any one has experience or know any open source client for AstManproxy? My main goal is to monitor every started and hung up call into a CDR, but with particular features: - Every call is started via AMI with Originate command. - I wanna keep record of the brigded call and both calls: - Call party A - Call duration into my database - then call party B and bridge it with A and keep CDR of the call duration between A and B. Does any of you has experience with this? Best regards, Marco Mouta ___ --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] Memory leak res_perl
Hi, I'm using res_perl with asterisk 1.0.0. And after running asterisk a couply of months, I see that the process asterisk take a lot on memory. And asterisk will freeze. If I look in the logging I see that the last command asterisk perfomed is a call to a perl program. So I think that there is a memory link in res_perl. Does anybody know if this is the case and maybe knows a sollution to this problem? Kind regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede email: [EMAIL PROTECTED] internet: www.mobillion.nl ___ --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] FXS: No ringtone
On Jul 10, 2006, at 1:23 AM, yusuf wrote: Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before Oh yeah, what you are talking about is ring back, not ringtone. I think the r option in the asterisk dial command might help you as that forces ringback. Marty ___ --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] setting up an email to fax with asterisk
Hehe, ok. Thanks, On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: A different job PaulH On Mon, 2006-07-10 at 03:34 -0500, Erick Perez wrote: We used? what are you doing different now? On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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: RE: [asterisk-users] Asterisk with ISDN Fritz PCI card
Hi, Maybe this is dirty but this is how I did it (with capi but you can probably do it with anything you want): ***Suppress the Hisax drivers in conflict with capi: [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.fcpcipnp.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.fcpcipnp.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn_bsdcomp.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn_bsdcomp.ko.old ***Download http://www.avm.de/ftp/cardware/fritzcrd.pci/linux and make, make install ***move the newly created modules to the good place from /lib/modules/2.6.12-10-386/extra/ to /lib/modules/2.6.12-10-386/kernel/drivers *** add capi and fcpci to /etc/modules (now when you reboot your machine the modules are loaded) then ***apt-get the libraries for capi # apt-get install libcapi20-dev ***download chan_capi on ftp://ftp.chan-capi.org/chan-capi and make,make install, make install_config but this probably works with misdn or anything else. Tell me if this works or if it doesn't (I'm on ubuntu not debian but this should be almost the same) Good luck, Ben - Original Message - From: Guy Corbaz [EMAIL PROTECTED] Date: Sunday, July 9, 2006 2:03 pm Subject: RE: [asterisk-users] Asterisk with ISDN Fritz PCI card Hi, Thank you for the suggestion. I tried to use mISDN first, then CAPI and now I'm trying I4L. As I'm using Debian, I can not load the FRITZ drivers. I got the source from the official site and recompiled it, but there is a strange message in the log and the capi drivers are not loaded. The problem is more linked to drivers that Asterisk. If you have any tips to get this up and running, I would be very happy as my search on the Internet didn't allowed me to solve that issue. Bests regards, Guy. At 11:25 09.07.2006 +1000, you wrote: What are you using (misdn, capi, something else?) and what problems are you having? I submitted a patch recently to mISDN which should have fixed a problem on hangup, if that's the problem you are having then try the latest cvs mqueue branch of mISDN. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- users- [EMAIL PROTECTED] On Behalf Of Guy Corbaz Sent: Saturday, 8 July 2006 23:59 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with ISDN Fritz PCI card Dear all, I'm desperately trying to get Asterisk working with a FRITZ PCI card on Debian with kernel 2.6.15. I'm wondering if anybody has such a working installation. Thank you for your help, Guy. Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --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 Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --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
Re: [asterisk-users] Memory leak res_perl
On Mon, Jul 10, 2006 at 10:59:20AM +0200, Arjan Kroon wrote: Hi, I'm using res_perl with asterisk 1.0.0. And after running asterisk a couply of months, I see that the process asterisk take a lot on memory. And asterisk will freeze. If I look in the logging I see that the last command asterisk perfomed is a call to a perl program. So I think that there is a memory link in res_perl. Does anybody know if this is the case and maybe knows a sollution to this problem? Note that since Asterisk 1.0.0 many leaks in asterisk itself have been fixed. Asterisk has a build-time option of memory allocations debugging, which may help to trace a leak inside asterisk code. Do try to upgrade to latest stable, or maybe at least to 1.0.11.1 . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Libpri + Zaptel + Asterisk polycom_acd_functions error message
After using the trunk versions as below, it all compiled ok, and the polycom acd is working great, but the music on hold and meetme will now work. I do not have any digium cards, is the ztdummy installed with the truck version? Or is there any thing I need to change? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 04 July 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? You need the /trunk versions of libpri and zaptel instead of the branches/1.2 releases. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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
RE: [asterisk-users] Memory leak res_perl
Thanks, Can you maybe give me an example of such a build-in option sebuuging. Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: maandag 10 juli 2006 11:19 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Memory leak res_perl On Mon, Jul 10, 2006 at 10:59:20AM +0200, Arjan Kroon wrote: Hi, I'm using res_perl with asterisk 1.0.0. And after running asterisk a couply of months, I see that the process asterisk take a lot on memory. And asterisk will freeze. If I look in the logging I see that the last command asterisk perfomed is a call to a perl program. So I think that there is a memory link in res_perl. Does anybody know if this is the case and maybe knows a sollution to this problem? Note that since Asterisk 1.0.0 many leaks in asterisk itself have been fixed. Asterisk has a build-time option of memory allocations debugging, which may help to trace a leak inside asterisk code. Do try to upgrade to latest stable, or maybe at least to 1.0.11.1 . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Error on dial_exec_full
Hi list, I get this message sometimes ( randomly ) when queues are calling agents: Jul 10 11:26:46 ERROR[8856]: app_dial.c:1481 dial_exec_full: Could not stop autoservice on calling channel I'm trying to see where it comes from ... Does someone has an idea ??? Thanks in advance ! ___ --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] setting up an email to fax with asterisk
Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. This is incorrect. Check out http://iaxmodem.sourceforge.net Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Encrypting the Conversation
Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Zeeshan ___ --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] Which Fax Solution really works on IAX or SIP?
Hi, I am trying to setup fax on my phone system. Which fax-to-email and email-to-fax solution really works on IAX and SIP? Zeeshan ___ --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] AGI tutorials
Rizwan Hisham wrote: Anybody who knows a good source of AGI tutorials on the net? plz share How about the Asterisk Wiki? Flynn ___ --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] AGI tutorials
It really depends on the programming language you plan to use. I'd have a look at the PHPAGI first, but there is not much as to AGI per se as with the underlying programming language on one side and understanding Asterisk on the other Hope this helps l. On Mon, 10 Jul 2006 10:31:58 +0200, Rizwan Hisham [EMAIL PROTECTED] wrote: Anybody who knows a good source of AGI tutorials on the net? plz share -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Encrypting the Conversation
Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] Certain fax types cause problems
Hi, I was wondering whether anyone has any input into the reliability of faxing (over a PRI) using spandsp and rxfax. 99% of times this is a reliable combination - we use it almost exclusively, but there seem to be certain fax devices which have problems talking to us. Most notably fax modems, and a couple of HP multi-function devices. I have enabled full tracing of these problem devices, and generally find that they will not train at the lowest level, but sometimes will manage to send 90% of a page before failing. Any pointers on how to diagnose or improve this would be appreciated. Regards, Steve ___ --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] asterisk to mobile phone
On Mon, Jul 10, 2006 at 03:14:27PM +0800, Sam Tam wrote: Have a look at cyber-telecom.net. CT-GSM-1000 seems to be one of the cheapest GSM Gateway that you can buy right now. Which is biz, and Sam works for Cyber-Telecom ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Cisco 7941/7961/7971 wont register with asterisk
I believe that these sip phones only work with cisco call manager. Only the 7950 and 7960 have an open sip stack Per Møller wrote: After google’ing extensively, I now have sip firmware (8.0.2SR1/8.0.3) running on the 7941, 7961 and 7971 and I even have a SEP.cnf.xml that seems to have everything and works (thanks to articles on www.voip-info.org). BUT the phones do not register correctly with asterisk. Everything is correct in the SIP communication - the phone does a register, asterisk replies with a 401 Unauthorized, and then the phone sends another register which should contain an line like: Authorization: Digest username=201,realm=my.asterisk.dom... etc This line is missing on the 7941/7961/7971. Anybody know why? // Per ___ --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 -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group|Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: [EMAIL PROTECTED] | TELEPHONE: +61 7 3365 8220 | ___ --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] Re: Asterisk and NFS
My understanding is that asterisk will read the file before it is finished being written. The proper method for NFS would be to write into another folder on the same file system, then have a script move the file to the proper call file location. The script would run every 5 seconds or so. The script shouldn't have an issue with a half written file because the file move should fail if the file is still locked. This works because moving a file on the same file system is immediate. (no bits copied, just a FS pointer change) -- -- Steven http://www.glimasoutheast.org Kyle Hagan [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Any one have an idea why asterisk would ignore a callout file that was dropped in via nfs The permossions are the same as a file dropped in locally that works fine. Kyle -- CONFIDENTIALITY NOTICE: This message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. ___ --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] Re: Re: Feasability of using * forsmallappartmentbuilding?
And you might as well sell them Internet access as well, because some of them may try to use dialup Internet over that VOIP connection, which will fail miserably. (see all of the fax threads for reference.) -- -- Steven http://www.glimasoutheast.org Cory Andrews [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] For 12 households, you could probably get a business class DSL or Cable broadband internet connection and use it for Voice. Then get something like SIP trunking in place and maintain a few analog POTS lines for local calls and 911 considerations. You could have an Asterisk PBX routing local calls out the PSTN, and LD out the SIP trunks. Put an FXO card in your Asterisk server to connect the analog POTS lines. Use an external FXS gateway to connect your analog phones. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of augustynr Sent: Friday, July 07, 2006 1:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Feasability of using * for smallappartmentbuilding? Cory, Channel banks on the fxs side makes sense what about fxo? The same? I cannot bring pri as it is too expensive. Thanks, Read this topic online here: http://forum.globalvoicenet.com/viewtopic.php?p=1561#1561 ___ --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
RE: [asterisk-users] Choppy MOH (Cisco gateway)
Actually this seems to have fixed it!! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) You will also want to add no vad to your dial-peer config to disable voice activity detection. I do not think it will resolve your issue, but worth a shot. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Gibbs Sent: Sunday, July 09, 2006 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into
[asterisk-users] Re: Metermaid phone compatibility
The metermaid changes in head are very different, but there is a working 1.2.7.1 patch in the bug tracker. http://bugs.digium.com/view.php?id=5779 I believe that the 1.2.7.1 patch also works with 1.2.9.1. -- -- Steven http://www.glimasoutheast.org Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Interesting... will this patch (metermaid) work with 1.2.7 asterisk? On 7/7/06, shadowym [EMAIL PROTECTED] wrote: I have been experimenting with the new metermaid application that allows phones to monitor the status of a parked call using BLF. Does anyone know what BLF feature the phone needs to support to make this work. Is it basically the same as the Bristuff Devstate()? Anyone know which phones do and do not support this (metermaid, not Bristuff)? Of course SNOM seems to be the main one but there must be others. Sounds like perhaps the Polycoms work with it as well. The reason I am asking is that I have an Aastra 9133i with v1.4 firmware and I can't get it to work with metermaid or Devstate(). Aastra tech support phoned me about my Bristuff Devstate() question to them and indicated their phone does not support that with current firmware but they are looking at it for a future release. That answers my Devstate() question. The phone/firmware supports BLF monitoring of SIP extensions just fine. Someone on the bug issue in question http://bugs.digium.com/view.php?id=5779 stated they had their Aastra working with metermaid just fine so I am wondering if I am missing something here. The 480i and 9133i are both pretty much the same in terms of BLF support so which model I have shouldn't matter.Still scratching my head over the person who posted that and I don't know their email to confirm. Maybe he is heredimitripietro?? I am pretty sure I have it set up right. My GXP2000 seems to work with metermaid ok but show hints only shows the GXP2000 monitoring the call parking extension (701). Ie. It only shows one extension monitoring and since the GXP2000 is working that must be the one. I have a second extension configured and it is the Aastra 9133i. Of course I tried a few different settings in the Aastra GUI and messed around with the Asterisk config. ___ --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 ___ --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] Encrypting the Conversation
Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. Agreed. I have seen and heard of a lot of attempts to bring SRTP support into Asterisk but the idea of SRTP just doesn't make sense to me. Asterisk, and VoIP servers in general, are meant to be communications services not security services. In my mind at least, it would seem to make sense to let security hardware such as a router or firewall handle such tasks as encryption and let the phone server handle what it does, signaling and transcoding. Otherwise, you end up with a device that is not ever going to be optimized for security, handling your security. On top of that, you also are reducing the level of scalability you can achieve on the phone server by adding yet another chore to its duty roster. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] B2BUA Webbased and Click 2 dial apps
Hi Marnus, That is a good idea, I didnt think of thatJ thanks Dinesh. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk Sent: Thursday, July 06, 2006 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps Also have a look at .call files. You web app can just create a .call file and then move it to the right location and asterisk will place the call No manager interface needed. Marnus van Niekerk Opportunity is missed by most people because it isdressed in overalls and looks like work.Thomas Alva Edison - Inventor of 1093 patents,including the light bulb, phonogram and motion pictures. Dinesh wrote: Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
RE: [asterisk-users] B2BUA Webbased and Click 2 dial apps
I am not for the billing part, as its sip based, and its educational calls only. I mean between sip.edu community and my educational institute. So practically any sip uri should be able to be dialed from the website. I dunno I am just asking the ideas for the group. Regards, Dinesh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Friday, July 07, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps It would be hard to bill all this calls, if you are using dialout call files instead of Asterisk Manager API no ? How would you colect the call duraction of both call legs? Thks, Marco Mouta On 7/6/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Also have a look at .call files. You web app can just create a .call file and then move it to the right location and asterisk will place the call No manager interface needed. Marnus van Niekerk Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. Dinesh wrote: Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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
RE: [asterisk-users] Choppy MOH (Cisco gateway)
And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! Bill -Original Message- From: Bill Gibbs Sent: Monday, July 10, 2006 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) Actually this seems to have fixed it!! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) You will also want to add no vad to your dial-peer config to disable voice activity detection. I do not think it will resolve your issue, but worth a shot. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Gibbs Sent: Sunday, July 09, 2006 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are
Re: [asterisk-users] Encrypting the Conversation
Has anyone here tried to use zphone with SIP soft phones and Asterisk? Michael On Mon, 10 Jul 2006 07:35:34 -0400, Raymond McKay wrote: Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. Agreed. I have seen and heard of a lot of attempts to bring SRTP support into Asterisk but the idea of SRTP just doesn't make sense to me. Asterisk, and VoIP servers in general, are meant to be communications services not security services. In my mind at least, it would seem to make sense to let security hardware such as a router or firewall handle such tasks as encryption and let the phone server handle what it does, signaling and transcoding. Otherwise, you end up with a device that is not ever going to be optimized for security, handling your security. On top of that, you also are reducing the level of scalability you can achieve on the phone server by adding yet another chore to its duty roster. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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
Re: [asterisk-users] Certain fax types cause problems
On 7/10/06, Doug Lytle [EMAIL PROTECTED] wrote: Any pointers on how to diagnose or improve this would be appreciated. Install HylaFAX and iaxmodem on your Asterisk box. Thanks, I will do. I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to asterisk? Any downsides/gotchas to this that I should be aware of? Regards, Steve ___ --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] Encrypting the Conversation
Maybe in Asterisk 1.4 SecureRTP application would do that. Regards Henry J. Cobb escribió: Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. ___ --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] Encrypting the Conversation
Hi Raymond, Raymond McKay wrote: Agreed. I have seen and heard of a lot of attempts to bring SRTP support into Asterisk but the idea of SRTP just doesn't make sense to me. Asterisk, and VoIP servers in general, are meant to be communications services not security services. In my mind at least, it would seem to make sense to let security hardware such as a router or firewall handle such tasks as encryption and let the phone server handle what it does, signaling and transcoding. Otherwise, you end up with a device that is not ever going to be optimized for security, handling your security. On top of that, you also are reducing the level of scalability you can achieve on the phone server by adding yet another chore to its duty roster. I would have to strongly disagree - if Asterisk was toted as a kid's toy, and sold by Fisher Price, then maybe security has no importance. But, if Asterisk or any other VoIP platform, for that matter, is to be introduced into the enterprise, it *has* to provide security. Tapping a hard phone line requires physical access to it - tapping a VoIP line can be done from anywhere in the world, if the server is not secure enough. Just use the Monitor() command, and setup a cron job to compress to mp3 and upload to an FTP server, and you have the perfect tap. It can even discriminate callers, called numbers and extensions, which conventional taps cannot! That is at the server iself - you could then argue that the transit RTP could be tapped by a corrupt tech working for your ISP or provider, which could happen also with physical lines, the difference being that the RTP tap is so virtual it can be made to leave no trace. A physical tap can be found by a routine inspection on the lines, an RTP tap cannot. If we want Asterisk to be a step forward in the right direction, security concerns *must* be addressed at some stage. Setting up a VPN and other security measures are fine, but they won't protect you from certain forms of tapping or compromise. Besides, if you put the onus of encryption on RTP, it can be made part of the standard and become universal. Otherwise, will your organization's VPN be compatible with mine? Best regards, Mike ___ --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] Libpri + Zaptel + Asterisk polycom_acd_functions error message
You may need to recompile now that you've got zaptel/ztdummy installed so that your install sees that the proper zaptel exists now. On 7/10/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: After using the trunk versions as below, it all compiled ok, and the polycom acd is working great, but the music on hold and meetme will now work. I do not have any digium cards, is the ztdummy installed with the truck version? Or is there any thing I need to change? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 04 July 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? You need the /trunk versions of libpri and zaptel instead of the branches/1.2 releases. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] channel bank log
Hello Can any one may send me log when channel bank is work Best regards Viktor Tatianin ___ --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] NuFone suggests to use Vonage!!!!
Joe Baptista wrote: On Sun, 9 Jul 2006, Andrew D Kirch wrote: To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Some people are into SM I guess. We have used NuFone. No problems during that period. If you know what you doing it's not bad. regards joe ___ --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 heh.. yah.. i know someone who has been on the receiving end of that too.. signature.asc Description: OpenPGP digital signature ___ --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] Libpri + Zaptel + Asterisk polycom_acd_functionserror message
Is this correct: zaptel: make clean; make; make install asterisk: make clean; make; make install Will this recompile everything needed? I tried, but the meetme app still does not get compiled (and no music) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 10 July 2006 13:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message You may need to recompile now that you've got zaptel/ztdummy installed so that your install sees that the proper zaptel exists now. On 7/10/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: After using the trunk versions as below, it all compiled ok, and the polycom acd is working great, but the music on hold and meetme will now work. I do not have any digium cards, is the ztdummy installed with the truck version? Or is there any thing I need to change? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 04 July 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? You need the /trunk versions of libpri and zaptel instead of the branches/1.2 releases. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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
Re: [asterisk-users] Encrypting the Conversation
Are you talking about ZiPhone a USB device ? Mike Simple Simon http://www.simplesimon.com - Original Message - From: Michael Graves To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, July 10, 2006 6:44 AM Subject: Re: [asterisk-users] Encrypting the Conversation Has anyone here tried to use zphone with SIP soft phones and Asterisk?MichaelOn Mon, 10 Jul 2006 07:35:34 -0400, Raymond McKay wrote: Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN.Agreed. I have seen and heard of a lot of attempts to bring SRTP support into Asterisk but the idea of SRTP just doesn't make sense to me. Asterisk, and VoIP servers in general, are meant to be communications services not security services. In my mind at least, it would seem to make sense to let security hardware such as a router or firewall handle such tasks as encryption and let the phone server handle what it does, signaling and transcoding. Otherwise, you end up with a device that is not ever going to be optimized for security, handling your security. On top of that, you also are reducing the level of scalability you can achieve on the phone server by adding yet another chore to its duty roster.Regards,Raymond McKayPresidentRAYNET Technologies LLChttp://www.raynettech.com(860) 693-2226 x 31Toll Free (877) 693-2226 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Call-limit and internal transfer
Hi, I set the sip.conf parameter call-limit=1 to limit outbound calls and 'disable' call waiting. But internally, I want to enable transfers. If the call-limit=1, the transfers fails. Any help ? Thanks all, Alexandre ___ --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] Encrypting the Conversation
On Mon, 2006-07-10 at 07:34 -0500, Mike Bates wrote: Are you talking about ZiPhone a USB device ? Mike zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike SRTP this does not rely on the server itself to provide the encryption. It also lets you be reasonably assured that if the numbers displayed match then not only is no one listening now, but they havent since you paired both endpoints. There is a drawback that SRTP can solve however, zphone only works on voip networks where the media proxy does not alter the data stream, it cannot be used to bridge to different channel types and codecs. This means that if you want to call out on the PSTN, SRTP can encrypt over the internet where zphone cannot. So it has its benefits, but also its drawbacks. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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] Unable to configure my DID number
Hi friends,At present, I am making outgoing calls using Teliax service with Asterisk. But, I am unable to receive calls. My DID number is: 3031234567. I am using SIP Server (Asterisk) setup, which is provided on Teliax website support. I have replaced my DID number i.e., 3031234567 in YOURNUMBER. But, I am unable to receive calls. My configuration file in extensions.conf File:exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)exten = 3031234567,1,Answer()exten = 3031234567,1,DIAL(SIP/user,20)---I hope the above configuration is proper, If not please suggest the modifications.In addition, I have some doubts.1) How should I configure my DID number in extensions.conf file to recevice incoming calls?2) Are they any
modifications required in "Features" option in my account on Teliax website?3) To receive incoming calls, do I need to make any kind of modifications to other configuration files in "Asterisk" and setup DID number?4) Do I need to set Public IP in my Asterisk server or our local IP is enough?4) After configuring DID number, where can I receive the phone call (ring)?5) How can I setup IVR (Interactive Voice Response) system to my DID number. (i.e., If someone calls to my DID number, then our IVR (Welcome message) should respond and ask for extension number.)Please respond to this message ASAP. Looking forward to your response.Thank you.Regards,Chandra.
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RE: [asterisk-users] intel vs amd motherboards
I think it's the same, 10 calls in 200ms = 50 calls in 1s because 1s = 5 x 200ms IMHO, is better to use seconds as period, because is more ease to compare rate speeds of each codec that are in bits per second. fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de C F Enviado el: Domingo, 09 de Julio de 2006 06:57 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] intel vs amd motherboards Thanks for that Tzafrir. Why does it ignore the secend CPU? BTW, on a side note on this topic, how can one calculate simultaneous transcoded channels using show transalation? In the case where it tells me 17 ms for encoding and 4 for decoding, that gives me 21ms per channel, in what time frame can I squeeze in how many channels before the calls start becoming intolerable? In other words should I aim for a 200ms time frame which means that I will get around 10 channels? or can I aim for a full second? which will give me around 50 channels? Thank You On 7/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 ___ --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] Re: Metermaid phone compatibility
I can confirm that the 1.2.7.1 patch works with 1.2.9.1 as well. On 7/10/06, Steven [EMAIL PROTECTED] wrote: The metermaid changes in head are very different, but there is a working 1.2.7.1 patch in the bug tracker. http://bugs.digium.com/view.php?id=5779 I believe that the 1.2.7.1 patch also works with 1.2.9.1. -- -- Steven http://www.glimasoutheast.org Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Interesting... will this patch (metermaid) work with 1.2.7 asterisk? On 7/7/06, shadowym [EMAIL PROTECTED] wrote: I have been experimenting with the new metermaid application that allows phones to monitor the status of a parked call using BLF. Does anyone know what BLF feature the phone needs to support to make this work. Is it basically the same as the Bristuff Devstate()? Anyone know which phones do and do not support this (metermaid, not Bristuff)? Of course SNOM seems to be the main one but there must be others. Sounds like perhaps the Polycoms work with it as well. The reason I am asking is that I have an Aastra 9133i with v1.4 firmware and I can't get it to work with metermaid or Devstate(). Aastra tech support phoned me about my Bristuff Devstate() question to them and indicated their phone does not support that with current firmware but they are looking at it for a future release. That answers my Devstate() question. The phone/firmware supports BLF monitoring of SIP extensions just fine. Someone on the bug issue in question http://bugs.digium.com/view.php?id=5779 stated they had their Aastra working with metermaid just fine so I am wondering if I am missing something here. The 480i and 9133i are both pretty much the same in terms of BLF support so which model I have shouldn't matter.Still scratching my head over the person who posted that and I don't know their email to confirm. Maybe he is heredimitripietro?? I am pretty sure I have it set up right. My GXP2000 seems to work with metermaid ok but show hints only shows the GXP2000 monitoring the call parking extension (701). Ie. It only shows one extension monitoring and since the GXP2000 is working that must be the one. I have a second extension configured and it is the Aastra 9133i. Of course I tried a few different settings in the Aastra GUI and messed around with the Asterisk config. ___ --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 ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Certain fax types cause problems
Steve Davies wrote: I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to asterisk? Any downsides/gotchas to this that I should be aware of? No, iaxmodem gives HylaFAX software modes that can also communicate with Asterisk. From the iaxmodem home page: IAXmodem is a software modem written in C that uses an IAX channel (commonly provided by an Asterisk PBX system) instead of a traditional phone line and uses a DSP library instead of DSP hardware chipsets. IAXmodem was originally conceived to function as a fax modem usable with HylaFAX http://hylafax.sourceforge.net, and it does that well. However IAXmodem also has been known to function with mgetty+sendfax and efax. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Re: Out of Office Auto Reply:
On Friday, June 23, 2006 4:08 PM Steven wrote: Exchange changes http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp Looks promising and helps a bit. Still no use of precedence bulk etc. though. Very poor detection of lit mails. ___ --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] Unable to configure my DID number
Hi
first:
exten = 3031234567,1,Answer()
exten = 3031234567,2,DIAL(SIP/user,20)
if this still don't work try
exten = _3031234567,1,Answer()
exten = _3031234567,2,DIAL(SIP/user,20)
second:
You have in sip.conf [teliax] configured, did You specify context=
?
if yes, then all dialplan for incoming calls should be in that
context
if no, then [general] context= should contain dialplan for incoming
calls
IVR for diffrent DID
exten = _3031234567,1,Goto(IVRfor1stDID|s|1)
exten = _3031234568,1,Goto(IVRfor2ndDID|s|1)
exten = _3031234569,1,Goto(IVRfor3rdDID|s|1)
and [IVRforXXXDID] context should have playback/background and menu
options
-Filip
Użytkownik Crazy Boy napisał:
My configuration file in extensions.conf File:
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = 3031234567,1,Answer()
exten = 3031234567,1,DIAL(SIP/user,20)
---
I hope the above configuration is proper, If not please suggest the
modifications.
In addition, I have some doubts.
1) How should I configure my DID number in extensions.conf file to
recevice incoming calls?
2) Are they any modifications required in "Features" option in my
account on Teliax website?
3) To receive incoming calls, do I need to make any kind of
modifications to other configuration files in "Asterisk" and setup DID
number?
4) Do I need to set Public IP in my Asterisk server or our local IP is
enough?
4) After configuring DID number, where can I receive the phone call
(ring)?
5) How can I setup IVR (Interactive Voice Response) system to my DID
number. (i.e., If someone calls to my DID number, then our IVR (Welcome
message) should respond and ask for extension number.)
Please respond to this message ASAP. Looking forward to your response.
Thank you.
Regards,
Chandra.
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RE: [Asterisk-Users] H.264 and Asterik?
Haven't read this whole thread (got way behind in this list :) ) Polycom has a softphone with video support also. Not sure if it is good or not, just downloaded the trial version to test it out. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Saturday, May 20, 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H.264 and Asterik? Kevin: Thanks for the info, I think I will buy the video phones Erick W. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 19, 2006 6:18 PM Subject: Re: [Asterisk-Users] H.264 and Asterik? Erick Weber V. wrote: Dose someone know if the latest version of asterisk support H.264? Asterisk SVN trunk (which will become Asterisk 1.4) supports H.264, and I have a Grandstream H.264 phone on my desk right now which I am testing with it (and it works fine!). ___ --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 ___ --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] outgoing call problem
Hi,
I have configured digium tdm04b card with asterisk
on debian. Incoming call is ok. But outgoing call has problem. Would you give me
advice ?
Here is my config files.
zaptel.conf
fxsks=1fxsks=2fxsks=3fxsks=4
loadzone=usdefaultzone=us
zapata.conf
[channels]language=en
context=incoming
signalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=nousecallingpres=yescallwaitingcallerid=yestransfer=noechocancel=yesechocancelwhenbridged=yesechotraining=yesrxgain=1txgain=4group=1callgroup=1pickupgroup=1immediate=nomusiconhold=defaultbusydetect=yescallprogress=nochannel = 1-4
extension.conf
[general]static=yeswriteprotect=no
[home]
exten = s,1,Answerexten =
s,3,Playback(thank-you-cooperation)exten = s,4,WaitExten
exten =
_1XXX,1,Playback(thank-you-cooperation)exten = _1XXX,2,Answerexten
= _1XXX,3,Wait(1)exten =
_1XXX,4,Playback(thank-you-for-calling)exten =
_1XXX,5,Dial(SIP/${EXTEN},10)exten =
_1XXX,8,Voicemail(u${EXTEN})exten = _1XXX,9,Hangupexten =
_1XXX,103,Voicemail(b${EXTEN})exten = _1XXX,104,Hangup
exten = _9.,1,Answerexten =
_9.,1,Playback(thank-you-cooperation)exten =
_9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()exten =
s,2,Background(/tmp/greetings);exten =
s,2,Background(enter-phone-number10)exten =
1,1,Playback(digits/1)exten = 1,2,Goto(sumiya,s,1)exten =
2,1,Playback(digits/2)exten = 2,2,Goto(ganbaa,s,1)exten =
i,1,Playback(pbx-invalid)exten = i,2,Goto(incoming,s,1)exten =
t,1,Playback(vm-goodbye)exten = t,2,Hangup( )
[sumiya]exten =
s,1,Dial(SIP/1001,10)exten = s,2,Hangup
[ganbaa]exten =
s,1,Dial(SIP/1000,10)exten = s,2,Hangup
Regards,
Ganbaa
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RE: [asterisk-users] Unable to configure my DID number
Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:
exten = 4567,1,Answer()
exten = 4567,1,DIAL(SIP/user,20)
Hope this helps.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, July 10, 2006 7:45
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable
to configure my DID number
Hi friends,
At present, I am making outgoing calls using Teliax service with Asterisk. But,
I am unable to receive calls. My DID number is: 3031234567. I am using SIP
Server (Asterisk) setup, which is provided on Teliax website support. I have
replaced my DID number i.e., 3031234567 in YOURNUMBER. But, I am unable to
receive calls.
My configuration file in extensions.conf File:
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = 3031234567,1,Answer()
exten = 3031234567,1,DIAL(SIP/user,20)
---
I hope the above configuration is proper, If not please suggest the
modifications.
In addition, I have some doubts.
1) How should I configure my DID number in extensions.conf file to recevice
incoming calls?
2) Are they any modifications required in Features option in my
account on Teliax website?
3) To receive incoming calls, do I need to make any kind of modifications to
other configuration files in Asterisk and setup DID number?
4) Do I need to set Public IP in my Asterisk server or our local IP is enough?
4) After configuring DID number, where can I receive the phone call (ring)?
5) How can I setup IVR (Interactive Voice Response) system to my DID number.
(i.e., If someone calls to my DID number, then our IVR (Welcome message) should
respond and ask for extension number.)
Please respond to this message ASAP. Looking forward to your response.
Thank you.
Regards,
Chandra.
How low will we go? Check out Yahoo! Messengers low PC-to-Phone
call rates.
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Re: [asterisk-users] outgoing call problem
I'm not a a guru, but
Check this line:
exten = _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten = _9.,2,Dial(Zap/g1/${EXTEN:1})
Hope it helps.
Ps. Give me some feedback if you solved the problem
On 7/10/06, Ganbaa [EMAIL PROTECTED] wrote:
Hi,
I have configured digium tdm04b card with asterisk on debian. Incoming call
is ok. But outgoing call has problem. Would you give me advice ?
Here is my config files.
zaptel.conf
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel = 1-4
extension.conf
[general]
static=yes
writeprotect=no
[home]
exten = s,1,Answer
exten = s,3,Playback(thank-you-cooperation)
exten = s,4,WaitExten
exten = _1XXX,1,Playback(thank-you-cooperation)
exten = _1XXX,2,Answer
exten = _1XXX,3,Wait(1)
exten = _1XXX,4,Playback(thank-you-for-calling)
exten = _1XXX,5,Dial(SIP/${EXTEN},10)
exten = _1XXX,8,Voicemail(u${EXTEN})
exten = _1XXX,9,Hangup
exten = _1XXX,103,Voicemail(b${EXTEN})
exten = _1XXX,104,Hangup
exten = _9.,1,Answer
exten = _9.,1,Playback(thank-you-cooperation)
exten = _9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()
exten = s,2,Background(/tmp/greetings)
;exten = s,2,Background(enter-phone-number10)
exten = 1,1,Playback(digits/1)
exten = 1,2,Goto(sumiya,s,1)
exten = 2,1,Playback(digits/2)
exten = 2,2,Goto(ganbaa,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(incoming,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup( )
[sumiya]
exten = s,1,Dial(SIP/1001,10)
exten = s,2,Hangup
[ganbaa]
exten = s,1,Dial(SIP/1000,10)
exten = s,2,Hangup
Regards,
Ganbaa
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--
Com os melhores cumprimentos,
Marco Mouta
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[asterisk-users] Dial command option D(digits)
I am using asterisk 1.2.9.1. I had been using the option D to send some dtmf tones after the call is answered. This doesnt seem to be working for me now. I am using and IAX2 connection from one machine to another. my extensions.conf has: exten= 57,1,Dial(IAX2/boxa_to_boxb/597,,tD(101)) When I answer phone 597, I do not hear the DTMF tones 101 being pulsed out. On the CLI I see Sending DTMF (101) to the called party. Am I not doing something correctly? Jerry ___ --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] IAX2 failed to authenticate as priv (DUNDi)
Hi all,I'm stuck here! I am trying to get DUNDi to work and it seems DUNDi is working accept the IAX part I think. I'm trying to let an extension from Trixbox1 call an extension on Trixbox2 with the use of DUNDi.
Ext 1301 * TrixBox1 * ---IAX2 * TrixBox2 *---Ext 1601 I got two trixboxes setup and a dundi lookup at the CLI works like a charmeTrixBox1asterisk1*CLI dundi lookup
[EMAIL PROTECTED] 1. 0 IAX2/priv:[EMAIL PROTECTED]/1601 (EXISTS|NOUNSLCTD|NOCOMUNSLTD) from 00:0c:29:30:77:fa, expires in 5 sDUNDi lookup completed in 35 msasterisk1*CLI dundi show peers
EID Host Model AvgTime Status 00:0c:29:30:77:fa 192.168.1.16
(S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0 unmonitored]
TrixBox2asterisk1*CLI dundi lookup [EMAIL PROTECTED] 1. 0 IAX2/priv:[EMAIL PROTECTED]/1301 (EXISTS|NOUNSLCTD|NOCOMUNSLTD) from 00:0c:29:0c:ab:c2, expires in 5 s
DUNDi lookup completed in 24 msasterisk1*CLI dundi show peersEID Host Model AvgTime Status 00:0c:29:0c:ab:c2
192.168.1.22
(S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0 unmonitored]But when I'm trying to make the call I get the following message:Jul 10 15:28:29 DEBUG[4389]
channel.c: Set channel IAX2/192.168.1.16:4569-1 to read format ulawJul 10 15:28:29 DEBUG[4389] channel.c: Set channel SIP/1601-fcaf to write format ulawJul 10 15:28:29 DEBUG[4389] channel.c: Set channel SIP/1601-fcaf to read format ulaw
Jul 10 15:28:29 DEBUG[4389] channel.c: Set channel IAX2/192.168.1.16:4569-1 to write format ulawJul 10 15:28:29 NOTICE[827] chan_iax2.c: Host
192.168.1.16 failed to authenticate as priv
Jul 10 15:28:29 WARNING[827] chan_iax2.c: Call rejected by 192.168.1.16: No authority foundJul 10 15:28:29 DEBUG[827] chan_iax2.c: Immediately destroying 1, having received reject
Jul 10 15:28:29 DEBUG[4389] channel.c: Hanging up channel 'IAX2/192.168.1.16:4569-1'Jul 10 15:28:29 DEBUG[4389] chan_iax2.c: We're hanging up IAX2/192.168.1.16:4569-1 now...Jul 10 15:28:29 DEBUG[4389] chan_iax2.c: Really destroying IAX2/192.168.1.16:4569-1 now...
I have the following config in my iax_custom.conf on both boxes:[priv]type=userdbsecret=dundi/secretcontext=dundi-priv-incomingdisallow=allallow=ulawallow=g726my dundi.conf
on trixbox1 looks like this:
[general]
port=4520
entityid=00:0C:29:0C:AB:C2
cachetime=5
ttl=32
autokill=yes
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nounsolicited,nocomunsolicit,nopartial
[00:0C:29:30:77:FA] ; TrixBox2
model = symmetric
host = 192.168.1.22
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
dynamic=yes
my dundi.conf on trixbox2 looks like this:[general]port=4520entityid=00:0C:29:30:77:FAcachetime=5ttl=32autokill=yes[mappings]priv = dundi-priv-local,0,IAX2,
priv:[EMAIL PROTECTED]
/${NUMBER},nounsolicited,nocomunsolicit,nopartial[00:0C:29:0C:AB:C2] ; TrixBox1model = symmetrichost =
192.168.1.22inkey = dundioutkey = dundiinclude = priv
permit = privqualify = yesdynamic=yesOn both boxes the extensions-custom.conf have the following contexts:; Macro Block;--
[macro-stdexten]; standard extension macroexten = s,1,Answerexten = s,2,Dial(SIP/${ARG1},25,t)exten = s,3,Goto(s-${DIALSTATUS},1)exten = s-NOANSWER,1,Voicemail(u${ARG1})exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b${ARG1})exten = s-BUSY,2,Hangupexten = _s.,1,Goto(s-NOANSWER,1)exten = a,1,VoicemailMain(${ARG1})[macro-dundi-lookup]; Goto the extension number. Check the local context first, followed by lookup
; dundi-priv-lookup is a pointer to the switch statement which will look for; extensions on other machines. This allows the convergence of multiple; Asterisk servers with different extension number blocks. Very cool!
;exten = s,1,Goto(${ARG1},1)include = dundi-priv-localinclude = dundi-priv-lookup; Directory Service Contexts;--
[dundi-test-canonical][dundi-test-local]; if we want to do a local lookup of numbers we advertise to dundi-test, then; we can just look hereinclude = dundi-test-canonicalinclude = dundi-pstn-local
[dundi-test-lookup]; if we want to lookup an external number, use this context. The switch; statement lets us search our peers for the number we are requestingswitch = DUNDi/dundi-test[dundi-pstn-local]
[dundi-priv-local]; we only have extensions 1600 - 1699 locallyexten = _16XX,1,Macro(stdexten,${EXTEN}) ;I got _13XX on my other box[dundi-priv-lookup]; Check our private peers for the exten #. Search 'priv' dundi context
switch = DUNDi/priv[dundi-priv-incoming]; when we get an incoming call from a private peer, it gets directed hereinclude = dundi-priv-local;--
; Outgoing Calls Contexts;--[local]; For extensions starting with 1300 - 1399 do a
[asterisk-users] QueuePauseMember(|Agent/) question
Hi List, Just a little question about QueuePauseMember() I use it in the manager with the following action: Action: Command\r\n Command: PauseQueueMember(|Agent/$id)\r\n\r\n; where $id is the agent's ID but the agent is still taking calls... Do I have to use the phone number agents are logged in instead of their ID ? Thanks for your reply in advance ! Tristan ___ --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] Sip No Audio Both Side
Hi I've setup asterisk as client, it was working very fine 2 months before, but now there is no audio on both side, as i'm on same network as provider. My IP = 111.111.20.* Provider Host IP = 111.111.10.10 sip.conf [general] context=sip-incoming bindport=5060 bindaddr=0.0.0.0 insecure=very srvlookup=yes disallow=all allow=iLbc nat=no register = mynum:[EMAIL PROTECTED]:4353 (not using 5060) i noticed CLI call comes in and when asterisk play a background(file) caller doesnt listen anything or even cannt pass dtmf to asterisk. I tried all codec but fails. Anyone help me out ? Fregi reg ___ --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: RE: RE: [Asterisk-Users] Very bad quality with AVMFritz!cardPCIandchan_capi
Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. Ben I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Ben Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James Unfortunately I can't, because the LAN board is integrated and I haveno PCI device left anymore. Ben You seem to have 2 wctdm adapters. Can you swap one of them with the fritz card? I'll try the usb trick first, and then if it doesn't work I'll try to swap one of the TDM400 with the fritz. But I can't do it now because people in my company won't be able to phone while I do that, which is completely impossible. Ben Yep. It's always the users causing problems! Good luck! James So I've just had the time to swap and disable usb in my bios and it changed nothing the quality is still the same (which means horrible). How could I check where the problem comes from? Ben ___ --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] QueuePauseMember(|Agent/) question
Sorry Misreading of the wiki, I have to use instead Action: QueuePause Interface: Agent/ID Pause: 1|0 I write it to the mailing list so that people will be aware of the way to use this manager command ;) Tristan a écrit : Hi List, Just a little question about QueuePauseMember() I use it in the manager with the following action: Action: Command\r\n Command: PauseQueueMember(|Agent/$id)\r\n\r\n; where $id is the agent's ID but the agent is still taking calls... Do I have to use the phone number agents are logged in instead of their ID ? Thanks for your reply in advance ! Tristan ___ --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
Re: [asterisk-users] FXS: No ringtone
Martin Joseph wrote: On Jul 10, 2006, at 1:23 AM, yusuf wrote: Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before Oh yeah, what you are talking about is ring back, not ringtone. I think the r option in the asterisk dial command might help you as that forces ringback. The r option seldom fixes ringback issues. Make sure you have /etc/asterisk/indications.conf setup. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] zaphfc - problem
hI, I've got problem with zaphfc kernel module. After I load into kernel i receive something like that into syslog: 0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 0). Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff, card = 0). Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 30, stat = 0xfc, card = 0). in my zapta.conf i'vh got: [channels] language=en signalling = bri_net_ptmp group = 1 context=default channel = 1 i've tried to change bri_net_ptmp into bri_cpe_ptmp but it didn't help. ISDN card is connected with good cable (i've plugged out phone) - or should I use ISDN crossover cable? I've searched google, but I didn't find anything helpfull. Regards, Marcin begin:vcard fn:Marcin J. Kowalczyk n:Kowalczyk;Marcin Janusz email;internet:[EMAIL PROTECTED] tel;work:+48 501 522 511 tel;home:+48 71 722 80 22 note;quoted-printable:skype:kowalma=0D=0A= gg:1171510 x-mozilla-html:FALSE version:2.1 end:vcard ___ --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] loading graphic on a Cisco 7960
I have an image in CIP format that i'm trying to load onto a 7960 phone using sccp. I don't know where to reference the data in which config file. Any help available? -- Edward F. Klimowicz Voicenet Systems Administration [EMAIL PROTECTED] 215.259.2131 ___ --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] zaphfc - problem
Hi Marcin, Marcin J. Kowalczyk wrote: hI, I've got problem with zaphfc kernel module. After I load into kernel i receive something like that into syslog: 0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 0). Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff, card = 0). Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 30, stat = 0xfc, card = 0). in my zapta.conf i'vh got: [channels] language=en signalling = bri_net_ptmp group = 1 context=default channel = 1 i've tried to change bri_net_ptmp into bri_cpe_ptmp but it didn't help. ISDN card is connected with good cable (i've plugged out phone) - or should I use ISDN crossover cable? I've searched google, but I didn't find anything helpfull. Regards, Marcin If you want to connect a telephone to the HFC card you need a crossed cable to connect to an NTBA to which you connect the phones. As far as I know you don't need a crossed cable when you connect the phone directly. Have you set HFC mode correctly (NT or TE)? Kai ___ --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] multiple calls
Title: Message I want to allow a SIP caller to place multuiple consecutive calls - So a caller connects to Asterisk and gets routed to a destination with the Dial command. After the call completes I would like to let them optionally enter a new destination. Currenty the call always disconnects - I assume becuase the SIP BYE is transmitteed to the caller. Thanks gc ___ --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] I need help patching source
I'm trying to provide dial tone on EM Wink type trunks. I found where in source, 'chan_zap.c' where I believe the code needs to be added. Basically I believe I can copy parts used for PRI in to EM and EM Wink signal types. However with my attempts, it fails to compile at chan_zap. And I'm not sure how to proceed now. It seems there should be a way to only recompile 'chan_zap.so' without doing a full recompile - Any help would be appreciated for this novice. Thanks, Bart ___ --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] Which Fax Solution really works on IAX or SIP?
Zeeshan Zakaria wrote: I am trying to setup fax on my phone system. Which fax-to-email and email-to-fax solution really works on IAX and SIP? Due to the way that your question has been asked I assume that you're talking about IAX or SIP over a latent internet connection to some VoIP provider... or between your own Asterisk systems through latent internet connections. And in that case the answer is that there is no fax solution that will really work. If you want fax to really work then you need to pass the audio over a medium that will not corrupt that audio before it gets to the receiver. Run-of-the-mill internet-strewn VoIP connections are not suitable. Lee. ___ --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] Re: Metermaid phone compatibility
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine. Still curious if anyone has this working on an Aastra phone? I can't get it to work but someone in the bug.digium.com list said they had it working on an Aastra phone. Maybe I am missing something. I tried just about everything I can think of. How does metermaid work? Is it using devicestate() similar to what the bristuff patch does or is it a different mechanism. What does the phone need to support in order for this to work. As far as I know, Aastra phones only support SIP device monitoring for BLF with the current firmware. -Original Message- From: Steven [mailto:[EMAIL PROTECTED] Sent: Monday, July 10, 2006 4:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Metermaid phone compatibility The metermaid changes in head are very different, but there is a working 1.2.7.1 patch in the bug tracker. http://bugs.digium.com/view.php?id=5779 I believe that the 1.2.7.1 patch also works with 1.2.9.1. -- -- Steven http://www.glimasoutheast.org Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Interesting... will this patch (metermaid) work with 1.2.7 asterisk? On 7/7/06, shadowym [EMAIL PROTECTED] wrote: I have been experimenting with the new metermaid application that allows phones to monitor the status of a parked call using BLF. Does anyone know what BLF feature the phone needs to support to make this work. Is it basically the same as the Bristuff Devstate()? Anyone know which phones do and do not support this (metermaid, not Bristuff)? Of course SNOM seems to be the main one but there must be others. Sounds like perhaps the Polycoms work with it as well. The reason I am asking is that I have an Aastra 9133i with v1.4 firmware and I can't get it to work with metermaid or Devstate(). Aastra tech support phoned me about my Bristuff Devstate() question to them and indicated their phone does not support that with current firmware but they are looking at it for a future release. That answers my Devstate() question. The phone/firmware supports BLF monitoring of SIP extensions just fine. Someone on the bug issue in question http://bugs.digium.com/view.php?id=5779 stated they had their Aastra working with metermaid just fine so I am wondering if I am missing something here. The 480i and 9133i are both pretty much the same in terms of BLF support so which model I have shouldn't matter.Still scratching my head over the person who posted that and I don't know their email to confirm. Maybe he is heredimitripietro?? I am pretty sure I have it set up right. My GXP2000 seems to work with metermaid ok but show hints only shows the GXP2000 monitoring the call parking extension (701). Ie. It only shows one extension monitoring and since the GXP2000 is working that must be the one. I have a second extension configured and it is the Aastra 9133i. Of course I tried a few different settings in the Aastra GUI and messed around with the Asterisk config. ___ --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 ___ --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: Re: [asterisk-users] Help with MusicOnHold
Hi Everyone, I was wondering if anyone had any ideas regarding this. I can see in the sip debug that music on hold is called when a person is put on hold, however I hear nothing. Any help would be appreciated. Thanks Julian From: [EMAIL PROTECTED]To: [EMAIL PROTECTED]; asterisk-users@lists.digium.comSubject: RE: Re: [asterisk-users] Help with MusicOnHold!!!Date: Fri, 7 Jul 2006 19:21:24 +CC: Hi Alyed,I used the rpm from Mandriva and it was labelled 1.1. The asterisk version in this RPM is actually 1.0.8. I also set the musiconhold=default no change in behavior.ThanksJulian Date: Fri, 7 Jul 2006 11:37:28 -0700Subject: RE: Re: [asterisk-users] Help with MusicOnHold!!!From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.comCC: 2 things might worth having a look:a) set up in your zapata.conf: musiconhold=defaultb) You say the asterisk version is 1.1, but 1.1 is developement version, maybe was just a typo, but you should be using either a 1.0.X or 1.2.X versionAlyed Return-Path: [EMAIL PROTECTED] Fri Jul 07 10:53:31 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Fri, 7 Jul 2006 10:53:31 -0700Hi,Yes we have that installed. Music on hold works fine when called in extensions.conf e.g. exten = 5000,2,MusicOnHold()However when I put someone on hold the music does not playI am using the Polycom Soundstation IP301 and X-lite phones.ThanksJulian Date: Fri, 7 Jul 2006 09:35:56 +0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with MusicOnHold Doyouhavethe"mpg123"utilityinyoursystem?Bydefault,Asterisk uses"mpg123"toplaythemp3filesformusiconhold. -kokmeng. JulianVaraniniwrote: Hi, Iamrunningasterisk1.1.Whenaclientisplacedonholdfromthe x-liteorpolycomphone,noholdmusicisheard.Ihave musicclass=defaultsetupinsip.confanddefaultexistsin musiconhold.conf.Hasanyonehadasimilarexperience?Anyhelpwould beappreciated. Thanks Julian ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: 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] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
Hi, Has anyone used the cooker RPM for asterisk version 1.2.9? I would like to hear some feedback before I install it. Thanks Julian From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Metermaid phone compatibility Date: Mon, 10 Jul 2006 09:02:37 -0700 Yes,Iamusingthe1.2.7.1patchon1.2.9.1.Itseemedtoworkfine. StillcuriousifanyonehasthisworkingonanAastraphone?Ican'tgetit toworkbutsomeoneinthebug.digium.comlistsaidtheyhaditworkingon anAastraphone.MaybeIammissingsomething.Itriedjustabout everythingIcanthinkof.Howdoesmetermaidwork?Isitusing devicestate()similartowhatthebristuffpatchdoesorisitadifferent mechanism.Whatdoesthephoneneedtosupportinorderforthistowork. AsfarasIknow,AastraphonesonlysupportSIPdevicemonitoringforBLF withthecurrentfirmware. -OriginalMessage- From:Steven[mailto:[EMAIL PROTECTED] Sent:Monday,July10,20064:30AM To:asterisk-users@lists.digium.com Subject:[asterisk-users]Re:Metermaidphonecompatibility Themetermaidchangesinheadareverydifferent,butthere isaworking1.2.7.1patchinthebugtracker. http://bugs.digium.com/view.php?id=5779 Ibelievethatthe1.2.7.1patchalsoworkswith1.2.9.1. -- -- Steven http://www.glimasoutheast.org"Matt"[EMAIL PROTECTED]wroteinmessage news:[EMAIL PROTECTED] Interesting...willthispatch(metermaid)workwith1.2.7asterisk? On7/7/06,shadowym[EMAIL PROTECTED]wrote: Ihavebeenexperimentingwiththenewmetermaidapplicationthat allowsphonestomonitorthestatusofaparkedcallusing BLF.Does anyoneknowwhatBLFfeaturethephoneneedstosupportto makethis work.IsitbasicallythesameastheBristuff Devstate()?Anyone knowwhichphonesdoanddonotsupportthis(metermaid,not Bristuff)?OfcourseSNOMseemstobethemainonebut theremustbe others.SoundslikeperhapsthePolycomsworkwithitaswell. ThereasonIamaskingisthatIhaveanAastra9133iwithv1.4 firmwareandIcan'tgetittoworkwithmetermaidorDevstate(). AastratechsupportphonedmeaboutmyBristuffDevstate() question tothemandindicatedtheirphonedoesnotsupportthat withcurrent firmwarebuttheyarelookingatitforafuturerelease. ThatanswersmyDevstate()question. Thephone/firmwaresupportsBLFmonitoringofSIP extensionsjustfine. Someoneonthebugissueinquestion http://bugs.digium.com/view.php?id=5779statedtheyhad theirAastra workingwithmetermaidjustfinesoIamwonderingifIam missingsomethinghere. The480iand9133iarebothprettymuchthesameinterms ofBLFsupportso whichmodelIhaveshouldn'tmatter.Stillscratching myheadoverthe personwhopostedthatandIdon'tknowtheiremailtoconfirm. Maybeheishere"dimitripietro"?? IamprettysureIhaveitsetupright.MyGXP2000seemstowork withmetermaidokbut"showhints"onlyshowstheGXP2000 monitoring thecallparkingextension(701).Ie.Itonlyshowsoneextension monitoringandsincetheGXP2000isworkingthatmustbe theone.I haveasecondextensionconfiguredanditistheAastra9133i.Of courseItriedafewdifferentsettingsintheAastraGUI andmessed aroundwiththeAsteriskconfig. ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: 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] OT: 3Com 3C10222 POE 24 Port Ethernet
There is an old, very old document that I found somewhere that this PoE switch was designed for NBX phones at that time. Does anybody in this list is using this switch with non-3com NBX PoE phones? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] zaphfc - problem
Kai Fürstenberg napisał(a): If you want to connect a telephone to the HFC card you need a crossed cable to connect to an NTBA to which you connect the phones. i'm connecting to ISDN-NTBA from T-Com. As far as I know you don't need a crossed cable when you connect the phone directly. Have you set HFC mode correctly (NT or TE)? I've tried with NT or TE (modprobe zaphfc modes=1 zaphfc modes=0 ) begin:vcard fn:Marcin J. Kowalczyk n:Kowalczyk;Marcin Janusz email;internet:[EMAIL PROTECTED] tel;work:+48 501 522 511 tel;home:+48 71 722 80 22 note;quoted-printable:skype:kowalma=0D=0A= gg:1171510 x-mozilla-html:FALSE version:2.1 end:vcard ___ --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] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
Julian Varanini wrote: Hi, Has anyone used the cooker RPM for asterisk version 1.2.9? I would like to hear some feedback before I install it. I haven't, I find it just to easy to compile it under Mandriva. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Re: Metermaid phone compatibility
shadowym wrote: Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine. Still curious if anyone has this working on an Aastra phone? I can't get it to work but someone in the bug.digium.com list said they had it working on an Aastra phone. Maybe I am missing something. I tried just about everything I can think of. How does metermaid work? Is it using devicestate() similar to what the bristuff patch does or is it a different mechanism. What does the phone need to support in order for this to work. As far as I know, Aastra phones only support SIP device monitoring for BLF with the current firmware. The metermaid-1.2.7.1.txt patch uses devicestate (AST_DEVICE_NOT_INUSE and AST_DEVICE_INUSE) and SIP subscribe/notify messages. If you can use hints to monitor the status of normal lines, then it should work for the parking slots too. See the Parking Lot Status / Access from the Programmable Buttons / LEDs section at http://www.voip-info.org/wiki/view/Asterisk+phone+snom for the procedure for setting it up with Snom 360s. Maybe it will help with your Aastra too... (The trunk code has something different apparently. I'm not sure where that is documented.) - Mike ___ --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: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
Hi Doug, I thought of that as well. I am not a total n00b with Mandriva, just enough to be dangerous. Do you have any walk throughs that I could use as a guide? Do you create an asterisk user for it to run under? What other software should I install in order for it to not only compile properly but also for asterisk to run. Thanks Julian Date: Mon, 10 Jul 2006 12:41:53 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9 JulianVaraniniwrote: Hi, HasanyoneusedthecookerRPMforasteriskversion1.2.9?Iwould liketohearsomefeedbackbeforeIinstallit.Ihaven't,IfinditjusttoeasytocompileitunderMandriva. Doug -- BenFranklinquote: "ThosewhowouldgiveupEssentialLibertytopurchasealittleTemporarySafety,deserveneitherLibertynorSafety." ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: 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] Mutiple Homes one asterisk box
I have a asterisk box up and running great. I have another house in my backyard that also wants to use my asterisk box. I am running trixbox now and have two POTS lines connected to digium TDM400P as well as 1 voip line for long distance. I would like to keep these two houses as seperate as possible (one POTS line for one house the other POTS for other house and share the VOIP line). What is the best way to go about doing this? Both houses will have Budgetone sip phones and share the same ethernet network. Can I install two instances of asterisk on the same box or is there a better way? Any 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
Re: [asterisk-users] Mutiple Homes one asterisk box
You can place the phones at each house in a different context. Trunks, too. On 7/10/06, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote: I have a asterisk box up and running great. I have another house in mybackyard that also wants to use my asterisk box. I am running trixbox now and have two POTS lines connected to digium TDM400P as well as 1voip line for long distance. I would like to keep these two houses asseperate as possible (one POTS line for one house the other POTS for other house and share the VOIP line). What is the best way to go aboutdoing this? Both houses will have Budgetone sip phones and share thesame ethernet network. Can I install two instances of asterisk on thesame box or is there a better way? Any suggestions? ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
Re: [asterisk-users] zaptel errors
Ariel Batista wrote: Justin Johnson wrote: Hi All, I have centOS 4.3 installed and have attempted to install asterisk separately. I have installed all the modules as suggested on Asterisk downloads, more (via SVN) However, on the zaptel install I am getting the following errors. centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h This doesn't work if you use a smp kernel ! So I use : sed -i s/rw_lock/rwlock/ /usr/src/kernels/2.6*/include/linux/spinlock.h But ensure that there is only one kernel-devel version, uninstall the one that isn't in use ! make[3]: *** [/usr/src/zaptel/torisa.o] Error 1 make[2]: *** [_module_/usr/src/zaptel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.9-34.0.1.EL-i686' make[1]: *** [linux26] Error 2 make[1]: Leaving directory `/usr/src/zaptel' make: *** [all] Error 2 Any one have any ideas how I can solve this? Thanks in advance, Justin /Mats -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --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] Mutiple Homes one asterisk box
Is that the standard way of doing things? I found a bunch of asterisk hosting providers in my search on the best way to do this. Is this what they are doing? On 7/10/06, Tom Lynn [EMAIL PROTECTED] wrote: You can place the phones at each house in a different context. Trunks, too. On 7/10/06, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote: I have a asterisk box up and running great. I have another house in my backyard that also wants to use my asterisk box. I am running trixbox now and have two POTS lines connected to digium TDM400P as well as 1 voip line for long distance. I would like to keep these two houses as seperate as possible (one POTS line for one house the other POTS for other house and share the VOIP line). What is the best way to go about doing this? Both houses will have Budgetone sip phones and share the same ethernet network. Can I install two instances of asterisk on the same box or is there a better way? Any 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 ___ --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
RE: [asterisk-users] Choppy MOH (Cisco gateway)
Yes that is correct. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, July 10, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote: And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! So was the fix to reconfigure your gateway to not use VAD? Just want to be clear... Marty ___ --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
Re: [asterisk-users] intel vs amd motherboards
But my question is, those that mean that it will take 1 second to convert 50 channels? if so do I get a 1 second latency when coverting 50 channels? On 7/10/06, Fabio [EMAIL PROTECTED] wrote: I think it's the same, 10 calls in 200ms = 50 calls in 1s because 1s = 5 x 200ms IMHO, is better to use seconds as period, because is more ease to compare rate speeds of each codec that are in bits per second. fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de C F Enviado el: Domingo, 09 de Julio de 2006 06:57 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] intel vs amd motherboards Thanks for that Tzafrir. Why does it ignore the secend CPU? BTW, on a side note on this topic, how can one calculate simultaneous transcoded channels using show transalation? In the case where it tells me 17 ms for encoding and 4 for decoding, that gives me 21ms per channel, in what time frame can I squeeze in how many channels before the calls start becoming intolerable? In other words should I aim for a 200ms time frame which means that I will get around 10 channels? or can I aim for a full second? which will give me around 50 channels? Thank You On 7/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 ___ --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] Keeping stable 1.2.9.1 updated with patches
I use stable 1.2.9.1 for my servers. How can I maintain my asterisk 1.2.9.1 updated with the patches produced for that release, in case a patch fits a need? what should I do in MANTIS to see patches applied to 1.2.9.1? While looking at MANTIS I just (?) saw one entry for Product build 1.2.9.1 however there are 100s other entries that seems interesting (like codec negotiations) but im not sure if they were commited to 1.2.9.1 Thanks, -- Erick Perez ___ --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] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
Julian Varanini wrote: Hi Doug, I thought of that as well. I am not a total n00b with Mandriva, just enough to be dangerous. Do you have any walk throughs that I could use as a guide? Do you create an asterisk user for it to run under? What other software should I install in order for it to not only compile properly but also for asterisk to run. The only thing I make sure to do, is install ALL the development libraries. Including the perl, pear, python and php libraries. No, I don't have any walk thoughs or guides. For Zaptel, libpri and Asterisk, along with the addons, just make clean;make;make install work for me. Nothing special. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Mutiple Homes one asterisk box
On Jul 10, 2006, at 10:48 AM, Andrew Niemantsverdriet wrote: Is that the standard way of doing things? I found a bunch of asterisk hosting providers in my search on the best way to do this. Is this what they are doing? Yes,l I think that's what contexts are for... I am also relatively new at this, and experimenting using the contexts for separate locations and separate users. This works although it takes a moment to understand it. You can also use separate prepaid accounts for the VOIP long distance calls... I don't really see that separate trunking is needed in you case, although I admit I a not clear on what he means by this... Since you have such a small amount of traffic I don't see it as a big deal... Marty ___ --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: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
I have not done a lot of compiling in Mandriva. Did you createa directory for it, e.g. /data/asterisk? Thanks Julian Date: Mon, 10 Jul 2006 15:23:22 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9 JulianVaraniniwrote: HiDoug, Ithoughtofthataswell.Iamnotatotaln00bwithMandriva,just enoughtobedangerous.DoyouhaveanywalkthroughsthatIcould useasaguide?Doyoucreateanasteriskuserforittorununder? WhatothersoftwareshouldIinstallinorderforittonotonly compileproperlybutalsoforasterisktorun. TheonlythingImakesuretodo,isinstallALLthedevelopment libraries.Includingtheperl,pear,pythonandphplibraries. No,Idon'thaveanywalkthoughsorguides. ForZaptel,libpriandAsterisk,alongwiththeaddons,justmake clean;make;makeinstallworkforme.Nothingspecial. Doug -- BenFranklinquote: "ThosewhowouldgiveupEssentialLibertytopurchasealittleTemporarySafety,deserveneitherLibertynorSafety." ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: 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
