RE : RE : RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found
Hi Chawki, I use a Debian Etch (testing branch) distro for my * box. Here, my zaptel modules are all in a zaptel folder, not in an extra. And the complete path owns the kernel name without any extension as yours. I am not sure of what to do... But, at your place, I will tempt two things : - copy your /lib/modules/2.6.8.1-12mdkcustom/extra (all the * concerned files) to a new folder /lib/modules/2.6.8.1-12mdksmp/zaptel and retempt to modprobe zaptel and all your necessary modules. - search if you have another folders branch from /lib/modules/ (tell us what you have here). Tell us what. Best Regards, Francois BERGERET, France. -Message d'origine- De : chawki hammoud [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 00:18 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found HI: It gives me this: Linux version 2.6.8.1-12mdksmp ([EMAIL PROTECTED]) (gcc version 3.4.1 (Mandrakelinux (Alpha 3.4.1-3mdk)) #1 SMP Fri Oct 1 11:24:45 CEST 2004 --- [EMAIL PROTECTED] wrote: What is the result of your cat /proc/version ? -Message d'origine- De : chawki hammoud [mailto:[EMAIL PROTECTED] Envoyé : vendredi 30 décembre 2005 23:21 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found Hi: I searched for zaptel.ko and i found it in lib/modules/2.6.8.1-12mdkcustom/extra ,is that the correct directory for zaptel.ko . ___ --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 __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] name that vendor...
Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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
RE : [Asterisk-Users] RE:problem with X100P card
Title: Message http://www.digium.com/index.php?menu=configuration RTFM ;-) Best Regards, Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tejas ShahEnvoyé: samedi 31 décembre 2005 06:04À: asteriskObjet: [Asterisk-Users] RE:problem with X100P card hi all, I wanted to knw whether it is possible to make call to analog phone (outbound call) using X100P card. I have only single piece of card. I m receiving call from analog phone properly,but cant make outbound call. If any one have a dialplan structure pls tell me.Thanks,Tejas Yahoo! for Good - Make a difference this year. ___ --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] Notifications when host fails qualify
Jonathan k. Creasy wrote: I am looking to be notified via email when a host fails it's qualify (is unreachable). I found this patch (http://bugs.digium.com/view.php?id=5372) but I wasn't sure if I could get that from it. Anyone else tried this? Yes, but it doesn't send e-mails. You need to write a script that connects to the manager interface (AMI) and reads the generated events when a host becomes unreachable. /O ___ --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] RPID Issue
Ray Van Dolson wrote: Posted this to -dev, but it may be more appropriate here as I haven't released my patches for it... I've run into a couple issues relating to RPID. I have an Asterisk 1.2.1 installation doing SIP for SPA-2002 and PAP2-NA ATA's. From the Asterisk box, we then do SIP to a VoIP provider who handles the SIP to PSTN translation for us. Pretty straight forward. I decided to try using the RPID features in 1.2.1. Enabled all the trustrpid directives and sendrpid as well. However, when I dial *81 number on my Sipura (*81 makes the call private) I get a fast busy back from Asterisk. Upon further investigation, it appears that Asterisk is saying the Sipura is unauthorized. This only happens when I try and block caller ID from the Sipura though. Dug around in the source a bit and it seems that Asterisk uses the contents of the From header to authenticate the ATA against. Normally (when making a non-CLID blocked call), the Sipura sends a from header like the following: From: ROY sip:[EMAIL PROTECTED];tag=cec0ff0080328e51o0 Authentication works fine in this case. However, when the caller dials *81, the from header looks like this: From: Anonymous sip:[EMAIL PROTECTED];tag=db61581ae353a8e1o0 I believe this is why authentication is failing. Now, is this incorrect behavior by my ATA? Seems like it should populate the From header no matter what. On the other hand, I see that the 5305715503.pw.digitalpath.net username is available in two other places in the initial INVITE: * The Contact header: Contact: Anonymous sip:[EMAIL PROTECTED]:5060 * The RPID header: Remote-Party-ID: ROY sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling So, what I gander is happening is that Asterisk is using the contents of the From header the first time around to generate the auth challenge stuff (nonce, etc) which is sent back to the ATA. The ATA then replies with the Proxy-Authorization field with the *correct* username (the 530571...). This doesn't match up with what was in the From field (Anonymous) and thus authentication fails. Correct? Maybe Asterisk should initially use the username in the Contact field to do authentication on? Or the RPID header if available? In any case, my solution was to modify check_user_full() and if an RPID header is available, I copy the username out of it into the of variable and authentication succeeds and the call works fine with or without *81. The fix works for me, but I have a feeling there's a more correct way to address this issue. I'd like to know if my Sipura is misbehaving, or if Asterisk should be looking somewhere other than the From field for authentication info. Asterisk should look somewhere else - in the auth header - for authentication info. However, this is not easy to fix in the current version of chan_sip. I've tried coding that before (chan_sip2) but it required too large changes at the time. We're currently planning a new generation of chan_sip that will have a different authentication scheme, not based on the from: header unless it's a local policy to require the From: header to be the same as the Digest auth user name. So to summarize: The Sipura is doing the right thing, but Asterisk can not handle it today, since Asterisk requires a From: user name. You need to disable the caller ID in Asterisk, not in the Sipura. /O ___ --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] Multiple Realm Definitions?
S McGowan wrote: Hate to keep asking, but I've not been able to find it covered online or in docs. I know you can define multiple domains in the sip.conf, but can you define multiple realms? For instance, I use a central server that handles a couple of area codes, and I would like to be able to have authentication realms such as areacode.hostname.domain. Anyone? No you can't, but it's a good input for the new chan_sip3 :-) For those of you that don't know what an authentication realm is: A userID and password is valid within a HTTP auth realm. (SIP uses HTTP digest authentication). A realm can cover one to several servers, but if it covers several servers all servers has to be able to authenticate all users within the realm. The realm ID is a string that has to be globally unique and therefore the recommendation is to use the DNS host name if the realm covers only one server or a domain if it covers multiple servers. In Asterisk, you can set *one* realm in sip.conf for your Asterisk PBX. By default this is asterisk which if you use it can't be considered as globally unique ;-) You can also define authentications for outbound calls based on realms, teaching Asterisk how to authenticate in various domains if challenged. /O ___ --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] name that vendor...
yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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] GXP-2000 any good with * ?
Michiel van Baak wrote: Hinting works fine for me with the latest firmware. What version are you running? We use 1.0.1.9 but the leds next to the speeddials wont use latest * and latest gxp firmware, have a look here on how to do it: http://www.voip-info.org/wiki/view/GXP-2000 ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) Thanks Peter ___ --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] Easiest way to use HFC-S?
Pisac wrote: I'm reading voip-info... and it's only confusing me: zaphfc, zapbri driver package, bristuff... So, what to download and install? If I install bristuff from junghanns.net, should I also install something else (patch)? What is (and where is) that zapbri driver package? Go to the junghanns.net page, get the latest bristuff. Unpack. You will find among other things a readme file explaining what to do. Do I need to use this download.sh script in bristuff? I already have working Asterisk (same version), so why to download and install again? Can I only patch sources and then compile? You can do a lot of things. Following the readme and doing it the way the developers envisioned installing the software will help you succeed much easier. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] name that vendor...
Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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] CAPI and *
Hi, * Armin Schindler wrote on 24.12.2005 (13:18): I suggest you use chan_capi-cm from sourceforge.net instead of old 0.3.5/0.4.0. And when installing a new version, remove old files from installation like app_capi* done that. Now I got a bunch of other problems. I don't think they're related to asterisk but to basic CAPI configuration. It turns out that capi configuration can be a nightmare :( Sascha -- [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
RE: [Asterisk-Users] Sip man in the middle
Hi Mike, This is wanted because using to ATA back to back creates a number of problems with echo. Also a delay for CID and problems with DTMF decoding. Keep everything digital is the way to go. Agreed. But before getting started with Asterisk, I posted a similar idea to the group; it was met with a quite cool reception, on and off-list. See http://lists.digium.com/pipermail/asterisk-users/2004-October/068932.html . I ended up avoiding Vonage and using multiple other providers. That said, I believe that many users of non-BYOD ITSPs would benefit from a proxy such as you describe. Unfortunately, I'm not aware of anyone that has implemented it yet. If you undertake such a project, IMO you should do it in Asterisk, or as a separate process that can run on the same machine as Asterisk, because many more people would use it and contribute to its development. --Stewart ___ --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] CAPI and *
On Sat, 31 Dec 2005, Sascha Andres wrote: Hi, * Armin Schindler wrote on 24.12.2005 (13:18): I suggest you use chan_capi-cm from sourceforge.net instead of old 0.3.5/0.4.0. And when installing a new version, remove old files from installation like app_capi* done that. Now I got a bunch of other problems. I don't think they're related to asterisk but to basic CAPI configuration. It turns out that capi configuration can be a nightmare :( CAPI itself is very easy, but the card drivers are different... What problems do you have ? Armin ___ --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] [Announcement] chan_capi-cm 0.6.2 released
Hi all, I just released version 0.6.2 of chan_capi-cm for Asterisk. The package and notes can be found at sourceforge: http://sourceforge.net/projects/chan-capi Armin PS: This version of chan_capi-cm is already part of OpenPBX trunk. ___ --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] Short code to replace the confirmation key
Are there any short codes to save into a key to replace the confirmation key like the tick key on the Snom phones? For example I want to programme a call pickup key which will have the confirmation automatically initiated without lifting the handset or waiting for time out? In short how do I emulate the tick key on a function key like *8 + tick key?? for call pickup. Looking forward from anyone who has an answer. Stephen ___ --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] Short code to replace the confirmation key
Are there any short codes to save into a key to replace the confirmation key like the tick key on the Snom phones? For example I want to programme a call pickup key which will have the confirmation automatically initiated without lifting the handset or waiting for time out? In short how do I emulate the tick key on a function key like *8 + tick key?? for call pickup. Looking forward from anyone who has an answer. Stephen ___ --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] How to check Queue Statistics
Hi everybody, I've made sales, marketing and technical-support queues. Now I also want to check performance of the agents and queues on regular basis. How can I check things like # of calls in a certain queue, # of calls answered, # of calls not answered, average wait time, most wait time, least wait time, who/when logged in, etc. Zeeshan A Zakaria ___ --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] name that vendor...
My apologies, Cory. I am mistaken. I was not trying to imply that Voipsupply.com supplies sucky equipment either. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Cory Andrews wrote: Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Peter Bowyer wrote: side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. No problem here. Using the 1.0.1.13 (very beta:)) also, synching with an internal time server on our network. (wich then syncs to pool.ntp.org) cheers ___ --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] name that vendor...
welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. [EMAIL PROTECTED] wrote: Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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 : RE : [Asterisk-Users] name that vendor...
Hi, On Sat, 2005-12-31 at 17:04 +0400, Vahan Yerkanian wrote: welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. we had the same issue, but the problem has been solved. just upgrade the firmware. matteo. ___ --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] name that vendor...
I believe that the Micronet firmwares authorize to have separate accounts for each different ports in SIP version. I will check this this next week at job and I will feedback you the results. Best Regards, Francois BERGERET, France. -Message d'origine- De : Vahan Yerkanian [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 14:04 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : RE : [Asterisk-Users] name that vendor... welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. [EMAIL PROTECTED] wrote: Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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] CAPI and *
Hi, * Armin Schindler wrote on 31.12.2005 (12:47): CAPI itself is very easy, but the card drivers are different... What problems do you have ? Every application (not only asterisk) complains that capi is not loaded. I do have two choices for my isdn card: AVM B1 PCMCIA and Eicon Diva Mobile V90. I prefer the last on, because I have two of them and want to connect a isdn phone to it. I think the AVM card isn't capable running in nt mode. So far all modules are loaded and there doesn't seem to be an error in /var/log/messages. My loaded modules: ,[ module_list ]- | divacapi 157188 0 | divas 69324 0 | divadidd 11584 2 divacapi,divas | kernelcapi 44320 7 b1pci,b1dma,b1pcmcia,b1,divacapi,capidrv,capi | b1pci 9472 0 | b1dma 14980 1 b1pci | b1pcmcia6528 0 | b1 21632 3 b1pci,b1dma,b1pcmcia | capidrv27572 0 | isdn 121196 1 capidrv | capi 16960 0 | capifs 5768 2 capi ` capiinfo shows the error 'capi not installed - No such device or adress (6)'. A google search brought some tips but they doesn't seem to be related to my problem. Most tips are for passive pci,iso or usb cards. When I shut down unloading capi complains about busy kernelcapi - not sure how to track this down. Kind regards and a happy new year, Sascha -- [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
Re: [Asterisk-Users] CAPI and *
On Sat, 31 Dec 2005, Sascha Andres wrote: Hi, * Armin Schindler wrote on 31.12.2005 (12:47): CAPI itself is very easy, but the card drivers are different... What problems do you have ? Every application (not only asterisk) complains that capi is not loaded. I do have two choices for my isdn card: AVM B1 PCMCIA and Eicon Diva Mobile V90. I prefer the last on, because I have two of them and want to connect a isdn phone to it. I think the AVM card isn't capable running in nt mode. So far all modules are loaded and there doesn't seem to be an error in /var/log/messages. My loaded modules: ,[ module_list ]- | divacapi 157188 0 | divas 69324 0 | divadidd 11584 2 divacapi,divas | kernelcapi 44320 7 b1pci,b1dma,b1pcmcia,b1,divacapi,capidrv,capi | b1pci 9472 0 | b1dma 14980 1 b1pci | b1pcmcia6528 0 | b1 21632 3 b1pci,b1dma,b1pcmcia | capidrv27572 0 | isdn 121196 1 capidrv | capi 16960 0 | capifs 5768 2 capi ` capiinfo shows the error 'capi not installed - No such device or adress (6)'. A google search brought some tips but they doesn't seem to be related to my problem. Most tips are for passive pci,iso or usb cards. When I shut down unloading capi complains about busy kernelcapi - not sure how to track this down. The open-source diva driver (divas) does not support the Eicon Diva Mobile, the Diva Server Cards are available only. I don't know if the drivers from Eicon do support this card. But anyway, I don't think this card is capable doing NT-mode. The error 'capi not installed' just means, that there is no card/driver registered which provides CAPI 2.0 interface. For example the divas driver, just loading it (and divacapi) does not provide a CAPI card. The card itself must be loaded with the firmware and started (divactrl load command). Armin ___ --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] How to set features.conf to change the hangup key.
I want to modify features.conf to set a different key to hang up call. Rather than the usual * key. I gather it involves some application map settings etc. Does anyone have a clue? I have read the docs but can hardly find any examples. Regards Obelix This message was sent using IMP, the Internet Messaging Program. ___ --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] CAPI and *
Hi, * Armin Schindler wrote on 31.12.2005 (15:26): The open-source diva driver (divas) does not support the Eicon Diva Mobile, the Diva Server Cards are available only. I don't know if the drivers from Eicon do support this card. But anyway, I don't think this card is capable doing NT-mode. The eicon driver itself doen't support it. So I can't use this card :( I removed the modules from getting loaded. At least the AVM card should be supported? If not, what card (it should be active because I need to run my laptop on a ISDN port that only active cards) should I go for? Thanks, Sascha -- [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
RE: [Asterisk-Users] How to set features.conf to change the hangup key.
In features.conf [featuremap] automon = *1 ; One Touch Record atxfer = *2 disconnect = *97 ; this is just an example Bogdan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obelix Sent: Saturday, December 31, 2005 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to set features.conf to change the hangup key. I want to modify features.conf to set a different key to hang up call. Rather than the usual * key. I gather it involves some application map settings etc. Does anyone have a clue? I have read the docs but can hardly find any examples. Regards Obelix This message was sent using IMP, the Internet Messaging Program. ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Peter, After upgrading to 1.0.1.13 I had some miscellaneous problems on one of my GXP-2000's--it would grab an IP address, but it wouldn't get the time/date, it wouldn't register, blah blah blah. I could access the web interface OK, so it wasn't a network issue (I don't think). Anyway...I ended up resetting to factory defaults and all is well now. Maybe try that? That has solved some other problems I've had as well. I dunno if ur familiar with the process, but it's kinda screwy, here's the info: http://www.grandstream.com/user_manuals/GXP2000.pdf scroll down to the very last page. Do the other phones on the same network get the time and date OK from time.nist.gov? -ross -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Saturday, December 31, 2005 4:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) Thanks Peter ___ --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] CAPI and *
On Sat, 31 Dec 2005, Sascha Andres wrote: Hi, * Armin Schindler wrote on 31.12.2005 (15:26): The open-source diva driver (divas) does not support the Eicon Diva Mobile, the Diva Server Cards are available only. I don't know if the drivers from Eicon do support this card. But anyway, I don't think this card is capable doing NT-mode. The eicon driver itself doen't support it. So I can't use this card :( I removed the modules from getting loaded. At least the AVM card should be supported? If not, what card (it should be active because I need to run my laptop on a ISDN port that only active cards) should I go for? The AVM should work, but as far as I know not in NT-mode. I don't have any knowledge about cards for laptops, sorry. Armin ___ --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] How to set features.conf to change the hangup key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]: Does this option work with Asterisk 1.07? I tried it and it didn't work In features.conf [featuremap] automon = *1 ; One Touch Record atxfer = *2 disconnect = *97 ; this is just an example Bogdan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obelix Sent: Saturday, December 31, 2005 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to set features.conf to change the hangup key. I want to modify features.conf to set a different key to hang up call. Rather than the usual * key. I gather it involves some application map settings etc. Does anyone have a clue? I have read the docs but can hardly find any examples. Regards Obelix This message was sent using IMP, the Internet Messaging Program. ___ --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 This message was sent using IMP, the Internet Messaging Program. ___ --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] Outputting human readable info on a VoIP call'squality?
not sure if im understading what you need, but have you checked rrdtool and cacti?On 12/30/05, S McGowan [EMAIL PROTECTED] wrote:Thanks for the info, that is actually what I'm doing, I am looking for the third party util to output the human readable listing so I can give it to a client.I currently dump RTP/SIP via Asterisk and a simple capture filter to reduce theamount of processing needed to do it. ThanksSKM-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of BJ WeschkeSent: Friday, December 30, 2005 5:02 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Outputting human readable info on a VoIP call'squality?On 12/30/05, S McGowan [EMAIL PROTECTED] wrote: Hello, Anyone know of a program that can analyse the RTP media stream and then output a human readable graph or other file? I'd like to be able to show jitter, difference, and if possible, echoes and other articfacts within a file of some sort. Ethereal can show you a graph, but cannot save it as a file for presentation to a client. Thank you for any help you may be able to offer. There was a patch in the bugtracker a while back that collected rtcpinformation on a call. I don't know if it made it to mainstream Asterisk (I don't think it has yet), but that's probably a decentstart to what you're looking for. Next steps of course would be totake that info, store it, and have a 3rd party util generate the infoyou're looking for. --Bird's The Word Technologies, Inc.http://www.btwtech.com/___--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-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] How to check Queue Statistics
On 12/31/05, Zeeshan [EMAIL PROTECTED] wrote: Hi everybody, I've made sales, marketing and technical-support queues. Now I also want to check performance of the agents and queues on regular basis. How can I check things like # of calls in a certain queue, # of calls answered, # of calls not answered, average wait time, most wait time, least wait time, who/when logged in, etc. From the Asterisk CLI you can do show queues and show agents. There are also a number of third party tools, free and not-free, to take information from Asterisk and present it in real-time and on a historical basis. -- 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
RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] =
{
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #, #,
builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect, *, *,
builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF },
};
In case you do not have this, good changes are that, in case you need badly
this feature, you will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work with Asterisk 1.07? I tried it and it didn't work
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is just an example
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the hangup
key.
I want to modify features.conf to set a different key to hang up call.
Rather than the usual * key. I gather it involves some application map
settings etc.
Does anyone have a clue? I have read the docs but can hardly find any
examples.
Regards
Obelix
This message was sent using IMP, the Internet Messaging Program.
___
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RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Is there a means of monitoring the call for that sequence of keys then hanging
up the call if they are detected?
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] =
{
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #, #,
builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect, *, *,
builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF },
};
In case you do not have this, good changes are that, in case you need badly
this feature, you will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work with Asterisk 1.07? I tried it and it didn't work
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is just an example
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the hangup
key.
I want to modify features.conf to set a different key to hang up call.
Rather than the usual * key. I gather it involves some application map
settings etc.
Does anyone have a clue? I have read the docs but can hardly find any
examples.
Regards
Obelix
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] How to set features.conf to change thehangup key.
Of course,
This is what res_features.c does...
B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Is there a means of monitoring the call for that sequence of keys then
hanging up the call if they are detected?
Indeed, this is 1.2.1
But do the following:
Go to the source tree, do a
vi res/res_features.c
Search for a :
struct ast_call_feature builtin_features[]
And you should see the builtin features:
In 1.2.1 I have:
#define FEATURES_COUNT (sizeof(builtin_features) /
sizeof(builtin_features[0]))
struct ast_call_feature builtin_features[] = {
{ AST_FEATURE_REDIRECT, Blind Transfer, blindxfer, #,
#, builtin_blindtransfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_REDIRECT, Attended Transfer, atxfer, , ,
builtin_atxfer, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_AUTOMON, One Touch Monitor, automon, , ,
builtin_automonitor, AST_FEATURE_FLAG_NEEDSDTMF },
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
In case you do not have this, good changes are that, in case you need
badly this feature, you will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change
thehangup key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work with Asterisk 1.07? I tried it and it didn't
work
In features.conf
[featuremap]
automon = *1 ; One Touch Record
atxfer = *2
disconnect = *97 ; this is just an example
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the
hangup
key.
I want to modify features.conf to set a different key to hang up call.
Rather than the usual * key. I gather it involves some application
map settings etc.
Does anyone have a clue? I have read the docs but can hardly find
any examples.
Regards
Obelix
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[Asterisk-Users] Having major issues with TDM2400
To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --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] name that vendor...
That's right, it's a welltech. I have one working but when people call in the ringing is not typical of American installations (indications?) and it freaks people out. Also, I don't get callerid. Where can I get the upgraded firmware? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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
[Asterisk-Users] voicemail/privacy system
Hello all. I'm relatively new to Asterisk, and before I get too involved in it, I want to find out if it will do what I'd like it to do (I'm relatively sure it can). In short, my goal is to set up a voicemail system and privacy manager for my home. For my proof of concept, I have a single port FXO card attached to a single POTS line. Currently, the FXO card sits in parallel to my phone. If I can get the answering/routing working the way I want, I'd upgrade that to a TDM11B and put my phone on the FXS side, completely hidden from the telco network. The privacy manager part is similar to the residental product many telcos are offering these days: when a call comes in with no caller-id, play a message stating that such calls aren't accepted and hang up. The voicemail part is also similar to what telcos are offering. When an incoming call (with caller-id, of course) comes in and isn't answered, it's routed to an attendant that allows the caller to press 1 to leave a message for Joe or 2 to leave a message for Jane, etc. So basically, the end product would be a single incoming line with a single physical extension and multiple virtual extensions (the voicemail boxes). Another feature I'd be interested in is being able to gathering up those voicemail messages, converting them to MP3 (if not already converted), and emailing them to the recipient (instead of leaving them in the voicemail system). I'm probably write that in perl with a mysql backend if there isn't already a tool out there for that. I currently have the kernel modules zaptel and wcfxo working and recognizing my card, as well as asterisk answering incoming calls and playing the demo, but I have no idea where to go from there. Any help would be appreciated. Thanks, Roy ___ --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] voicemail/privacy system
Yep, perfectly possible. I would do that with AGI and php, in your case, perl works as well. The only thing you need is read documentation regarding AGI, Voicemail and extensions. Its kind of difficult to helo you further if you dont tell us how much you know about contexts, extensions etc. But in general you will NEED to read about: 1. What are contexts, extensions, applications, macros, set, etc. 2. How to control the flow of a call using contexts logic 3. How to use AGI() to make things easier 4. It could be usefull to know how manager API works, so you can make things like setting user preferences to auto call back bridging the cell phone user with the person that just have leaved a message. Hope that helps. Best Regards Best RegardsOn 12/31/05, Morel Mosolff [EMAIL PROTECTED] wrote: Dear friends and business associates,I will be out of office until January the 12th, 2006. With kind regards,Morel Mosolff___--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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] wctdm module goes missing after a reboot - Gentoo?
do you have the folder /lib/modules/kernel-version-here/misc/ ?? and if so, you must have there the wcxxx drivers. If not, then trying to compile zaptel again and look for errors installing. good lookOn 12/30/05, Robert La Ferla [EMAIL PROTECTED] wrote: Moises Silva wrote: Hello Ryan. Check out the file /etc/modules.conf, /etc/modules.d/zaptel ... if for some reason you have empty the modules.conf, modules-update force will fix it, tough. In order to provide you with further help, please provide more clues.What about systems that use /etc/modprobe.conf?depmod should handle itbut it doesn't work when I tried it either but then again I haven't given this a lot of thought...___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] SIP Subscription Storage Location
On 12/14/05, Douglas Garstang [EMAIL PROTECTED] wrote: I guess I'm not fully grasping it because I don't see it as a problem. So asterisk gets stopped and started. The subscriptions are still there if they are cached. Why is that a problem? I think we would definitely make ourselves available for testing. What's the procedure for that? How do we let the powers that be know that we are interested in this feature, and would happily test it if available? For all those previously inquiring about it, bug 6047 in Mantis (http://bugs.digium.com) now has a pointer to some working code in it that allows sip subscriptions (eg. engine for BLF features on phones) to survive a 'reload' in Asterisk. More testers with different phones/setups are now needed to make sure no new problems are introduced with this change. Thanks. BJ -- 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
RE: [Asterisk-Users] How to check Queue Statistics
Where Can I get these free tools? Zeeshan A Zakaria -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Saturday, December 31, 2005 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to check Queue Statistics On 12/31/05, Zeeshan [EMAIL PROTECTED] wrote: Hi everybody, I've made sales, marketing and technical-support queues. Now I also want to check performance of the agents and queues on regular basis. How can I check things like # of calls in a certain queue, # of calls answered, # of calls not answered, average wait time, most wait time, least wait time, who/when logged in, etc. From the Asterisk CLI you can do show queues and show agents. There are also a number of third party tools, free and not-free, to take information from Asterisk and present it in real-time and on a historical basis. -- 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
[Asterisk-Users] misconfigured autoresponder
On Sat, 2005-12-31 at 19:44 +0100, Morel Mosolff wrote: Dear friends and business associates, I will be out of office until January the 12th, 2006. With kind regards, Morel Mosolff At the risk of causing yet another one of these can anything be done to whack this user from the list until Jan 12 when he returns? Misconfigured auto responders are bad m'kay This will double the volume to the list - at least he isnt creating a mail loop and responding to his response :/ Morel Mosolff [EMAIL PROTECTED] -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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
Re: [Asterisk-Users] How to check Queue Statistics
On 12/31/05, Zeeshan [EMAIL PROTECTED] wrote: Where Can I get these free tools? Zeeshan A Zakaria -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Saturday, December 31, 2005 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to check Queue Statistics On 12/31/05, Zeeshan [EMAIL PROTECTED] wrote: Hi everybody, I've made sales, marketing and technical-support queues. Now I also want to check performance of the agents and queues on regular basis. How can I check things like # of calls in a certain queue, # of calls answered, # of calls not answered, average wait time, most wait time, least wait time, who/when logged in, etc. From the Asterisk CLI you can do show queues and show agents. There are also a number of third party tools, free and not-free, to take information from Asterisk and present it in real-time and on a historical basis. AsteriskGuru Queue Statistics http://www.asteriskguru.com/tools/queue_stats.php I'm sure there are others. Maybe someone else can kick in a couple other links/projects? -- 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
RE: [Asterisk-Users] voicemail/privacy system
If you dont want to get too stuck into the guts of Asterisk yet, the [EMAIL PROTECTED] distribution can do all you have requested with a one button install web configuration via AMP. Personally I think its a great place to start with asterisk whatever your requirements as it makes a good base without having to go through the drudgery of installing asterisk the requirements/add-ons piecemeal, espically AMP, as the prereqs are a stress! (mumbles something about a, thankfully forgotten, nightmarish FreeBSD Asterisk/AMP install then fades into background, wimpering) :) Hope that helps, -Alex. -Original Message- From: Roy Kidder [EMAIL PROTECTED] Sent: 31/12/2005 18:41 To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [Asterisk-Users] voicemail/privacy system Hello all. I'm relatively new to Asterisk, and before I get too involved in it, I want to find out if it will do what I'd like it to do (I'm relatively sure it can). In short, my goal is to set up a voicemail system and privacy manager for my home. For my proof of concept, I have a single port FXO card attached to a single POTS line. Currently, the FXO card sits in parallel to my phone. If I can get the answering/routing working the way I want, I'd upgrade that to a TDM11B and put my phone on the FXS side, completely hidden from the telco network. The privacy manager part is similar to the residental product many telcos are offering these days: when a call comes in with no caller-id, play a message stating that such calls aren't accepted and hang up. The voicemail part is also similar to what telcos are offering. When an incoming call (with caller-id, of course) comes in and isn't answered, it's routed to an attendant that allows the caller to press 1 to leave a message for Joe or 2 to leave a message for Jane, etc. So basically, the end product would be a single incoming line with a single physical extension and multiple virtual extensions (the voicemail boxes). Another feature I'd be interested in is being able to gathering up those voicemail messages, converting them to MP3 (if not already converted), and emailing them to the recipient (instead of leaving them in the voicemail system). I'm probably write that in perl with a mysql backend if there isn't already a tool out there for that. I currently have the kernel modules zaptel and wcfxo working and recognizing my card, as well as asterisk answering incoming calls and playing the demo, but I have no idea where to go from there. Any help would be appreciated. Thanks, Roy ___ --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 This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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] Semi-OT: porting numbers away
Andrew Kohlsmith writes: An interesting wrinkle I'm running against is that you cannot port numbers from a cellular carrier to a landline. i.e. I can't port my cell # to a DID on my PRI. I am not sure if this is just a line of bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but I've not had the time to really dig in. They claim that between cell carriers numbers are portable but not from cell to landline. Some will do it, other will not. Key point is that they're not required to do so (by the FCC in the U.S., at least). http://www.fcc.gov/cgb/NumberPortability/ Interestingly, this official FAQ doesn't even contemplate porting from wireless to wireline. I have had Speakeasy VOIP decline to port my personal number because it was originally a cell number (from Verizon Wireless). However, Vonage did port it -- though it took them 3 months to do it. And, more recently, Broad Voice got it from Vonage, though it took them 6 months. ___ --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] Semi-OT: porting numbers away
On Sat, 2005-12-31 at 14:54 -0500, Janina Sajka wrote: However, Vonage did port it -- though it took them 3 months to do it. And, more recently, Broad Voice got it from Vonage, though it took them 6 months. Read the broadvoice user policy, if you port a number in only at their discretion can you port it out. If they decide they like your number, guess what, you cant have it back. Interesting concept on number portability. Because they arent a LEC they dont have to let you port it out... Gotta wonder about a company that puts something like that in their contract. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] OT Maybe: Anyone have any knowledge of v5.1/v5.2 in connection with Asterisk?
Hi All, I've been asked by a prospective client if Asterisk can is compliant with v5.1 and v5.2 - which I never heard about till today. After trying to figure out what I'm dealing with, it appears as some kind of signaling protocol, run on E1 lines. I was wondering if anyone has more information about this, in addition, if anyone knows any information about utilizing Asterisk in such a network - if at all. Regards, Nir S ___ --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] tdm dialout delay
Hello!
I am using asterisk 1.2.1 with a digium TDM card and trying to reduce the
dialout delay to 1/2 secs at the most, i could bring it down from 6/7
seconds to 3/4 seconds by tweaking the config and tone/zone/dtmf settings in
the source, still this is not acceptable as the regular pstn phone takes
less than 1 sec to ring on the called number
Dec 30 03:51:44 VERBOSE[13144] logger.c: -- Executing
Dial(SIP/1010-3211, Zap/g0/011234567) in new stack
Dec 30 03:51:44 DEBUG[13144] rtp.c: Channel 'Zap/1-1' has no RTP, not doing
anything
Dec 30 03:51:44 DEBUG[13144] chan_zap.c: Dialing '011234567'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 VERBOSE[13144] logger.c: -- Called g0/011234567
1 sec delay ?
Dec 30 03:51:45 DEBUG[13144] chan_zap.c: Exception on 16, channel 1
Dec 30 03:51:45 DEBUG[13144] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index 0)
3 secs delay ?
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Exception on 16, channel 1
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Enabled echo cancellation on
channel 1
Dec 30 03:51:48 VERBOSE[13144] logger.c: -- Zap/1-1 answered
SIP/1010-3211
[EMAIL PROTECTED] asterisk]# ztcfg -vvv
Zaptel Version: SVN-trunk-r880M
Echo Canceller: KB1
Configuration
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
4 channels configured.
zaptel.conf
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxsks=1
fxsks=2
fxsks=3
fxoks=4
loadzone= kr
defaultzone = kr
zapata.conf
[channels]
language=en
loadzone =kr
progzone =kr
signalling=fxs_ks
context=from-pstn
group=0
channel = 1
channel = 2
channel = 3
signalling=fxo_ks
context = from-internal
group=1
channel = 4
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
busydetect=no
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
extensions.conf entry
exten = s,1,Dial(Zap/g0/${EXTEN:1},20,tr)
exten = s,2,Hangup()
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[Asterisk-Users] New Manager Client Program
Here is a work-in-progress that provides pop-up note-taking windows based on caller-ID, outgoing call dialing from directory lookup selection, and other stuff. I hope it's useful to folks. http://asteroid.from.net ___ --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: Fax Support
I was able to get spandsp spandsp-0.0.2pre21.tar.gz working in 1.2.1, but you must manually copy the patch changes over to the Makefile in the \usr\src\asterisk\apps directory. I followed the following directions found by googling asterisk spandsp. === Make sure libtiff is installed on your machine. Versions 3.5.7, 3.6.0 and 3.7.1 seem to work OK. There have been several bugs related to FAX document handling in some recent versions of libtiff. Also, some people have had trouble using spandsp because they had more than one version of libtiff on their machine. Take care with this. You will also need libxml2 installed. The FAX facility does not use this, but some other parts of spandsp do. If you are using an RPM based system, such as RedHat or Fedora, you will need the libtiff, libtiff-devel, libxml2 and libxml2-devel RPMs installed. Use the usual: ./configure make make install process to build the spandsp library. Note that if you use configure in this way, the software will be installed in /usr/local. In this case make sure your /etc/ld.so.conf file has an entry for /usr/local/lib and then run 'ldconfig' command. Next, put app_rxfax.c, app_txfax.c and Makefile.patch in your Asterisk apps directory. Use the command: patch Makefile.patch within the apps directory, to patch your make file so it will build the new application. If the patching process fails, don't be too surprised. The patch file was generated for a specific revision of Asterisk, and things change. It would be difficult to produce a completely generic patch. If you look through the patch, and the Makefile, I think most people should be able to work out what is needed. Now rebuild and install Asterisk. (I had to manually insert the +lines data from the patch to the Makefile, and be sure to observe TABs as space char are not acceptable and halted the compiler.) Now if you put something like: exten = 1234567,1,rxfax(/home/steveu/testfax.tif) in your Asterisk extensions.conf file, a call to 1234567 should invoke the fax facility, to receive a fax to the file /home/steveu/testfax.tif. Alternatively: exten = 1234567,1,txfax(/home/steveu/testfax.tif) in your Asterisk extensions.conf file will cause a call to 1234567 to invoke the fax facility to send the file /home/steveu/testfax.tif to a calling fax machine. When sending a fax it is more likely you will be calling out to the remote FAX machine. In this case, make your Asterisk call the far FAX machine, and when it answers do: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) The addition of |caller will make txfax act as a calling machine, rather than an answering machine. - Original Message - From: [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Friday, December 30, 2005 1:17 PM Subject: Fax Support Can anyone guide me enabling fax support in asterisk. I tried spandsp patch but was unsuccessful. Because patch for chan_sip.c was not proper for asterisk's version 1.2.1. Can anyone help me adding fax support in asterisk 1.2.1. ___ --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] Need HT488 FXO example for both inbound and outbound.
I'm new to Asterisk and I'm looking for example of how to set up the FXO side of an HT488. I have the FXS side working and can place calls between it and soft phone just fine. What I was able to find the Wiki, forums google has not been useful to me. I think I'm missing something simple probably on the HT488 device. Once I have working example I'd be happy to post it on the Wiki for others. BTW, I purchased the HT488 because I was told it's a direct replacement for the Supra 3000 which is no longer directly available to end users per Cisco. If it's the HT488 that's a piece of junk someone please let me know so I can return it. Thanks James Ronald ___ --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: Fax Support
On Sat, 2005-12-31 at 16:45 -0600, LJ wrote:
exten = 1234567,1,rxfax(/home/steveu/testfax.tif)
I email pdf files, this requires a few extra packages, but I feel its
the easiest way to deal with it. I have a macro that calls a shell
script and all to do this. Its based off work of others, I dont know
who they are, but here is what I do. Its fairly trivial and fairly
painless if the underlying components are installed and working.
DEPENDS:
spandsp rxfax
libtiff (should include tiff2pdf)
mime-construct
everything else should be standard on an asterisk server
(bourne shell, asterisk, etc)
in AEL
macro faxreceive( email ) {
rxfax(/tmp/${UNIQUEID}.tiff);
system(/usr/sbin/mailfax /tmp/${UNIQUEID}.tiff ${email}
${CALLERIDNUM});
};
In extensions.conf
[macro-faxreceive]
exten = s,1,rxfax(/tmp/${UNIQUEID}.tiff)
exten = s,2,system(/usr/sbin/mailfax /tmp/${UNIQUEID}.tiff ${arg1}
${CALLERIDNUM})
Then just call it
in AEL
faxreceive([EMAIL PROTECTED]);
extensions.conf
exten = 12345,1,macro(faxreceive,[EMAIL PROTECTED])
Using a database or something you can dynamically get the email address
on a per user basis. Fairly trivial to do, course an AGI could do this
as well.
The mailfax script
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
PDFTMPFILE=${FAXFILE}.pdf
PDFFILE=`date +fax-%Y%m%d%H%M%S.pdf`
tiff2pdf -o /tmp/${PDFTMPFILE} -z -p letter \
-c creator such as your company name \
-a author such as your company name \
-t Fax from ${FAXSENDER} to ${RECIPIENT} \
-s Fax from ${FAXSENDER} to ${RECIPIENT} ${FAXFILE}
mime-construct --to $RECIPIENT \
--subject Fax from $FAXSENDER \
--attachment ${PDFFILE} --type application/pdf \
--file /tmp/${PDFTMPFILE}
rm -f /tmp/${PDFTMPFILE} /tmp/${FAXFILE}
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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[Asterisk-Users] Re: CALLERIDNUM (Rehan AllahWala)
Do u know how to instert it in the agi ?
$AGI-exec(SetCIDNum(8504338555));
but it didn't work
$AGI-exec('Set',CALLERID(number)=8504338555);
Freddi
www.voip-info.org/wiki-asterisk
or you could try the CLI show application Set, and show function
CALLERID
On 12/28/05, Rehan Ahmed [EMAIL PROTECTED] wrote:
Hi
Can you send any example of this command like
Set(CALLERID(num)=value)
Thanks
Rehan
On 12/28/05, C F [EMAIL PROTECTED] wrote:
in 1.2 and on (or CVS HEAD) you have to use:
Set(CALLERID(num)=value)
On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr
ser91623 cdr) exten = _00.,3,MYSQL(Query resultid
${connid} select\
if((floor(u.credit/p.cost))1\,ceil((u.credit)/p.cost)*60\,0)\
as\ sekund\ from\ user\ u\,\ sip\ s\,\ pricelist\ p\ where\
u.iduser=s.iduser\ and\ s.idsip=\'${CALLERIDNUM}\'\ and\
p.acode=s.acode\ and\ u.currency=p.currency\ and\
right(left(\'${EXTEN}\'\,CHAR_LENGTH(
p.ccode)+2)\,CHAR_LENGTH(p.ccode))\
like\ concat(p.ccode\,\'%\')\ order\ by\ p.ccode\ desc\ limit\
1) exten = _00.,4,MYSQL(Fetch foundRow ${resultid} sekund)
; fetch
row
..
..
without success. At row 3 have var ${CALLERIDNUM} original
value, not value from ${CALLNR}.
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[Asterisk-Users] SPA-3K FXO: Incoming and outgoing calls in different contexts?
I'm working to get my head around some of the conceptual underpinnings of SIP channels. In a couple of recent discussions, Kevin and others have noted that the notion of separate user and peer behavior from a SIP partner (for want of a better word) is not really germane, as it is in the world, say, of IAX. So I'm trying to apply that to the FXO port on my SPA-3000 with respect to its ability to both originate and terminate calls to/from the PSTN? I wonder what the cheeses would recommend. I want to be able to differentiate behavior based on the originate/terminate distinction, which is easy when one distinguishes between the user and peer entities in an iax.conf file. How would I do that if, as recommended, I have a single entry in the sip.conf file for the channel, defined as a peer? Thanks in advance for any good ideas out there, and a happy new year to all. . . B. ___ --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: Semi-OT: porting numbers away
trixter aka Bret McDanel [EMAIL PROTECTED] writes: Gotta wonder about a company that puts something like that in their contract. My favorite are the indemnification clauses. I count how many things some large company wants *me* to indemnify *them* against. Don't these jokers have a legal budget? Do they think any money I can chip in is going to amount to a hill of beans? In any case why would I want to agree to pay their legal expenses? (I'm not a lawyer so I might be misreading things a bit, but many of them sure seem to be very open-ended in what they want users to indemnify them against. They way I see it, if the user does something wrong that costs a company money, then the company can always sue the user. Indemnification clauses are simply a way to get money from the user even in cases when no court would agree with them that the user did something wrong.)) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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] Voicemail through outlook
Hello: I am using asterisk as my voicemail server for users coming through PSTN and SIP phones. I have asterisk integrated to an email server that handles emails from my users. I now want to give my users the facility to send voicemails from outlook client as well as the web interface they have to my email server. Any suggestions about how to develop an addon for outlook? As well as a voicemail plugin for the webinterface. Should I handle these voicemails at my email server end, or should I receive them at asterisk's end? Thanks for any suggestions/ideas/queries. Jami __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.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
