Re: [Asterisk-Users] Setting Variables
On Tue, 10 May 2005, Daniel Salama wrote: Is it possible to set a variable for an IAX device in iax.conf that can be read from the dial plan (extensions.conf)? If so, can you explain? Use setvar=name_of_var=value_you_want. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmware upgrade
Steve Prior wrote: I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Sounds like you got the one we just sent back to Sipura a couple weeks ago. Try to update the firmware again; if that fails I'd suggesting contacting Sipura for an RMA. The one we had seemed heat induced. It would work fine for a few minutes and then just die (but always still responded to pings). No calls and no web interface would work until it was unplugged and cooled down. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Avaya 4602 SIP phone
I tested Avaya 4602 a while ago and I successfully instaled SIP software and connected the phone to Asterisk. I remember that the upgrade procedure was a bit complicated (includes TFTP and Web server as I recall), but if you follow it to the letter you should have no problems. If the phone keeps rebooting, my guess is that it cannot connect to the Web server, or it cannot access the files on the server. However, this phone is designed specifically for proprietary Avaya platform, and I couldn't achieve anything but the most basic telephony functions. Therefore, I concluded that this phone is not suitable for Asterisk, so my suggestion is to try some other phone. Niksa Yao, Yuanbin wrote: Hi, I have been trying to connect Avaya 4602 SIP phone to Asterisk, but the phone keeps rebooting after I downloaded the SIP software (Avaya phone release 050205). I would like to know if anyone succeeded to hook up Avaya SIP phone to Asterisk. I appreciate your help! Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmwareupgrade
Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Prior Sent: May 11, 2005 2:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmwareupgrade I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. Thanks Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to detect a hang-up in the first 5 seconds
Hi, how can i detect a call which hang-up in the first 5 seconds. when i dial using sipgate.de without any money on my account i get an answer from sipgate and a hang-up after 3-5 seconds. So if they do so, i would like to choose the next provider. Any ideas?? bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream firmware 1.0.6.2
Hi, Don`t you know where can I download some older firmware ? I`m interested for latest stable one. Can anybody send me a 1.0.5.23 or anything else, which is good working ? Currently, I have version 1.0.5.18 and 1.0.5.11. Thanks, -b - Original Message - From: Michael D Schelin [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 4:39 AM Subject: Re: [Asterisk-Users] Grandstream firmware 1.0.6.2 this is beta code! I'm beta testing The t38. Don't use this unless your testing. It is not backwards compatible. Julio Arruda wrote: Doug Lytle wrote: Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ 2 quick notes, a quick test seem to indicate iLBC is broken (didn't try any troubleshooting). And, in the release notes, from what I remember, there are mentions of problems with dowgrading it, at least they recomend you to call support to do it) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
Gents Whilst I apreciate the sentiments regarding my question, if you are to look at my track record of helping people - across in the majority [EMAIL PROTECTED], AMP lists and to a lesser (but growing) extent asterisk-user and asterisk-biz, its not up there with the super gurus, but I am putting more back into the list. I have asked my fair share of questions as well, (and will continue to do so when needed). I noticed that you didn't make these comments when people have talked exclusivly about polycom or cisco hardware (as technically these are not asterisk either) There is a high technical aim to the list, there is also a community factor. I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Regards David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Freeworlddialup
Hi there i just setup my asterisk to dial with freeworlddialup and i am trying to dial 411 voice xml service from freeworlddialup and I always get congestion/busy Is this normal Also dialing the tell number hangs up on me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream firmware 1.0.6.2
In article [EMAIL PROTECTED], barney [EMAIL PROTECTED] wrote: Don`t you know where can I download some older firmware ? I`m interested for latest stable one. Can anybody send me a 1.0.5.23 or anything else, which is good working ? Currently, I have version 1.0.5.18 and 1.0.5.11. Visit http://gs-firmware.gratissip.dk/firmwares/ Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Log Output
Guys. Is there a way to output the same information shown on the console when invoked as - but to a log file for later grepping and such? I noticed the normal log only shows warning and errors but no info messages like in the console.. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone attached to Sipura SPA-1001 has no ring
On Tuesday 10 May 2005 02:52 pm, Malcolm Bader wrote: I hooked up a SPA-1001 with asterisk yesterday and all works well except the phone doesn't ring. The phone I'm using has a LCD display so I can see the call come in. (with caller id info) I can answer and complete the call but it's just not ringing. The phone rings if pluged into a POTS line so it's not the phone that's the problem. I've used the SPA-1001's web interface and the ringer seems to be enabled. Is there some setting I'm missing? Or is there something special I have to set up in asterisk? Sounds like what I had. My SPA-1001 came like this and was RMA'ed. The ring voltage is adjustable in the advanced settings. I tried both raising and lowering it to no avail. It wouldn't ring any of my phones. I don't recall seeing it generate callerid though, but to tell the truth I wasn't paying a lot of attention to that. The web server status page would say that the phone status was 'Ringing' while this was happening. And if I picked up the handset while this was happening, the calls would complete. -- Peter Wemm - [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] All of this is for nothing if we don't go to the stars - JMS/B5 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-addon
I had the same problem. you did a CVS checkout on the latest version download this version instead it fixed the problem for me. http://www.asterisk.org/html/downloads/asterisk-sounds-1.0.7.tar.gz -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Altus Snyman Verzonden: woensdag 11 mei 2005 7:44 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] asterisk-addon Good day all I downloaded asterisk-addons to try and make asterisk log in the sql db but when I make a make install i get this error cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Please help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Predictive Dialier
Hi Folks, Where can I find a list of Predictive dialer solutions for Asterisk? Thanks, Isamar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323
Hi I've a problem with the H.323 calls The Asterisk's version is 1.0.7 The version of pwlib library is 1.6.6 The openh323 library is 1.13.5 Oh323 driver channel's version 0.6.6 I use a gatekeeper openh323gk registered successfuly to asterisk when i do a call from client h323 connected to gatekeeper to sip client I've this signalling: Asteriskclient h.323=ohphone+phonejack setup Q.931 -- Ack Call proceeding Q.931 -- Ack -- At this point I must have a RAS message ARQ(Admission request) from Asterisk to gatekeeper,but I've Release complete Q.931 Have you suggestions? I think the problem is with libraries! Another question how can debug h323 channel? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Log Output
In /etc/asterisk/logger.conf, add this: full = notice,warning,error,debug,verbose Then watch /var/log/asterisk/full getting really big ;) Julian. On 5/11/05, Anton Krall [EMAIL PROTECTED] wrote: Guys. Is there a way to output the same information shown on the console when invoked as - but to a log file for later grepping and such? I noticed the normal log only shows warning and errors but no info messages like in the console.. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip or IAX2 eb Client
Is there any good IAX2 or SIP free web client? Im looking for something open source or free preferably IAX2 for integrating into a web site... Any leads? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P for UK
Can anyone tell me when the digium TDM400P will be certified for the UK? Also, are there any good modems that work well with * whithout caller id problems ( UK ) ? Initially, I would like to make a development machine with one FXO card ( TDM dev card if possible but modem would suffice ) Thanks in advance___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Predictive Dialers
Guys. Anybody know of any predictive dialers for Asterisk and Windows? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600
Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Predictive Dialers
Hi Anton, Start by having a look at this: http://www.voip-info.org/wiki-Predictive+dialer N -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 11 May 2005 10:19 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Predictive Dialers Guys. Anybody know of any predictive dialers for Asterisk and Windows? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600
On Wednesday 11 May 2005 11:23, Anton Krall wrote: I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Just in case you don't know, AS5350 supports SIP *and* H323 after IOS version 12.3 (maybe a little earlier). It allows you to use both at the same time, without needing to set it up for one system specifically. Haven't tried it with Asterisk yet though. Richard. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
David John Walsh wrote: Hello list we are installing 2 new servers (to run asterisk) shortly, for a stand alone service. Ignoring our current naming convention, we'd like to name them something.. but we are not sure what. I'd use constellations and star names. There are enough of them to ensure infinite scalability :) Other ideas: volcanos, greek mythology names, cyclons names... -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateway service under Asterisk
Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application successfully and I'm able to participate as an audio-only participant. What I want to do now is to include more than one teleconference room. When I connect to asterisk from a phone I want * to ask me for the room I want to connect to and for a password which should be read from a database and will be different for each room. Depending on the selected room, * should dial one sip address or another, which are read from a database as well. Please, note that I am not talking about Meetme rooms (although I don't know if I can archive my goal using it). I only want to dial a new SIP agent depending the selected room. How can achieve this? What additional tools will be necessary? Thanks a lot! Eduardo. Eduardo López Martínez [EMAIL PROTECTED] Isabel Operation Center [EMAIL PROTECTED] DIT - Dept. Ing. Sist. Telemáticos Tf: +34 913367366 (446) UPM - Univ. Politecnica de Madrid Fax: +34 913367333 Madrid SPAIN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600
Just in case you don't know, AS5350 supports SIP *and* H323 after IOS version 12.3 (maybe a little earlier). It allows you to use both at the same time, without needing to set it up for one system specifically. Haven't tried it with Asterisk yet though. I have tried it. I have SIP trunk between Asterisk and AS5300 (C3640 before), and it`s working good. It`s quite good solution, but its much more expensive as some PCI card direct in Asterisk (i`m using PRI interconnect to PSTN). -b PS: sorry for poor english On Wednesday 11 May 2005 11:23, Anton Krall wrote: I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Just in case you don't know, AS5350 supports SIP *and* H323 after IOS version 12.3 (maybe a little earlier). It allows you to use both at the same time, without needing to set it up for one system specifically. Haven't tried it with Asterisk yet though. Richard. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk starting problem
Hello Everybody, I am having problems with starting Asterisk. The message what I am getting is; May 11 15:41:32 WARNING[5031]: res_musiconhold.c:728 moh_scan_files: Cannot open [cdr_addon_mysql.so]May 11 15:41:32 WARNING[5031]: loader.c:305 __load_resource: libmysqlclient.so.10: cannot open shared object file: No such file or directory May 11 15:41:32 WARNING[5031]: loader.c:463 load_modules: Loading module cdr_addon_mysql.so failed! I have configured the modules.conf for loading the cdr_addon_mysql.so. But still the problem persists. If you could please help me to figure as to whats wrong, it would be very kind of you. Regards, Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gateway service under Asterisk
Hi Eduardo, Check the AGI (Asterisk Gateway Interface) scripting: http://home.cogeco.ca/~camstuff/agi.html http://www.voip-info.org/wiki-Asterisk+AGI If you write your own script that: * plays the welcome message * collects digits to select the conference room * redirects the call to the appropriate SIP address I think you will easily be able to solve your problem. Perhaps the list knows about other options, but this is how I guess I would do it. Good luck! --- Eduardo López Martínez [EMAIL PROTECTED] wrote: Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application successfully and I'm able to participate as an audio-only participant. What I want to do now is to include more than one teleconference room. When I connect to asterisk from a phone I want * to ask me for the room I want to connect to and for a password which should be read from a database and will be different for each room. Depending on the selected room, * should dial one sip address or another, which are read from a database as well. Please, note that I am not talking about Meetme rooms (although I don't know if I can archive my goal using it). I only want to dial a new SIP agent depending the selected room. How can achieve this? What additional tools will be necessary? Thanks a lot! Eduardo. Eduardo López Martínez[EMAIL PROTECTED] Isabel Operation Center [EMAIL PROTECTED] DIT - Dept. Ing. Sist. TelemáticosTf: +34 913367366 (446) UPM - Univ. Politecnica de Madrid Fax: +34 913367333 MadridSPAIN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip or IAX2 eb Client
Anton Krall wrote: Is there any good IAX2 or SIP free web client? Im looking for something open source or free preferably IAX2 for integrating into a web site... Any leads? Sounds like you're looking for the IAXClient libaries. There are many examples within it. This includes IAXCom, IAX2 ActiveX control etc. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Avaya 4602 SIP phone
Hi, I have been trying to connect Avaya 4602 SIP phone to Asterisk, but the phone keeps rebooting after I downloaded the SIP software (Avaya phone release 050205). I would like to know if anyone succeeded to hook up Avaya SIP phone to Asterisk. I appreciate your help! Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning of the Asterisk server
Hi, I am trying to hook up Avaya 4602 SIP phone to Asterisk server, but got the following warning: May 10 15:45:15 WARNING[2042]: Unexpected bind error: Cannot assign requested address May 10 15:45:15 WARNING[2042]: Unable to create RTP session: Cannot assign requested address May 10 15:45:15 WARNING[2042]: Unable to build sip pvt data for MWI Can somebody tell me what went wrong with the Asterisk configuration? Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not starting
Hi! I have problems with starting asterisk by command asterisk -c My OS is Mandrake 10.1 official and I Am using CAPI with AVM Fritz card and installing it was successful. I have attached lines after asterisk -c with all it's warnings first and also my modules.conf file. I'm pleased with any suggestions you got Thanks in advance! [EMAIL PROTECTED] asterisk]# asterisk -vc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.1-2mdk, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found -- This box has 1 capi controller(s). -- CAPI[contr1] supports DTMF -- CAPI[contr1] supports supplementary services HOLD/RETRIEVE TERMINAL PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR == ast_capi_pvt(50,*,demo,0,2) (1,2,64) == ast_capi_pvt(50,*,demo,0,2) (1,2,64) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD) [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Registered indication country 'us-o' -- Registered indication country 'gr' -- Registered indication country 'ru' -- Registered indication country 'nz' -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park'
Re: [Asterisk-Users] is it allowed to install 2 TE405P cards at same P.C.?
On May 11, 2005 12:31 am, Yousri Farouk wrote: Does Asterisk allow to install two pci TE405P Cards at the same P.C.? Yes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On May 10, 2005 09:13 pm, Paul wrote: How about lameass and question? This is the asterisk-users list. There are people on this list who need help with *important* things. I'm happy to see the odd discussion like this, instead of the usual What is the BARE MINIMUM I NEED to install Asterisk, Broadvoice is down again! and I don't want to pay for g729, where is the free version crap. Asterisk-users has no specific topic, aside from asterisk-related discussion. So yes, take your attitude and blow it right out your arse. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On May 11, 2005 03:38 am, David John Walsh wrote: I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Nonsense; it's little sidetracks like this that make the list interesting. You've done absolutely nothing wrong, and I for one am enjoying the different naming schemes (some old, some new) that are coming up here. Don't let the odd social miscreant scare you off. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Avaya 4602 / SIP response 481
Hi, I have been trying to connect Avaya 4602 SIP phone to Asterisk, but the phone keeps rebooting after I downloaded the SIP software (Avaya phone release 050205). I would like to know if anyone succeeded to hook up Avaya SIP phone to Asterisk. I appreciate your help! Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
Yeah, serves me right for writing e-mail while half awake. s/ATT/Bell Labs/g -- Black Ratchet blackratchet at blackratchet.org - http://www.blackratchet.org (914) 239-5132 x On Tue, 10 May 2005, trixter http://www.0xdecafbad.com wrote: On Tue, 2005-05-10 at 23:46 -0400, Black Ratchet wrote: My two toy boxes at work are 'succasunna', named after the 1st city to ever have #1ESS phone switch, and 'murrayhill', which was named after the location of the ATT headquarters. I wonder what that big building in basking ridge was then. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] forum www.asterisk-italia.it
Hi all! We just opened a forum dedicated to italian asterisk users. It's an attempt to offload some traffic on this huge traffic mailing list and to let italian users to help each other and share experiences. For all italian speaking users please visit and contribute to www.asterisk-italia.it! Thanks! Paolo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 Card ------ Adtran ------- FXS BUG???
Good Day list, I have googled and wiki'd myself to no avail. Here is my setup. Asterisk Box Digium T1 card connected to an adtran Total Access 624. I need to pass ANI and DNIS information from asterisk to the analog devices hanging off the adtran. The digits are announced to the fxs devices between ring 1 and ring 2 (If I pick up the anlog phone after ring 1, I can hear them so I feel good about this stage). Problem is that my analog device once it is finished with the call needs to be able to FLASHHOOKexten to transfer the call to a SIP Handset. Problem is that as soon as the FLASH (which is 700milli-seconds) is heard (and I can hear this by BARGING into the channel and listening) asterisk hangs up. I have tried all combinations of busydetect ( and and off) as well as every combination of busycount from 1 to 10. No help I have pulled out all my hair at this point. Weird thing is that if I change signaling to fxX_ks I am able to perform the transfer np, however I loose the ANI and DNIS (feature group d) between ring 1 and ring 2. ANY help would be appreciated. I think it may be a bug, but not sure. Ronald. PS I am happy to pay for services rendered provided a solution can be had. /etc/zaptel.conf span=1,1,0,esf,b8zs em=1-24 defaultzone=us loadzone=us /etc/asterisk/Zapata.conf language=en context=from-pstn faxdetect=none usecallerid=yes signalling=featd ; FXS for ringing phones echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 threewaycalling=yes transfer=yes rxwink=300 rxflash=700 busydetect=yes busycount=10 context=from-internal ; Points to the default context of your extensions.conf channel = 1-24 ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E M (Default) (Slaves: 01) Channel 02: E M (Default) (Slaves: 02) Channel 03: E M (Default) (Slaves: 03) Channel 04: E M (Default) (Slaves: 04) Channel 05: E M (Default) (Slaves: 05) Channel 06: E M (Default) (Slaves: 06) Channel 07: E M (Default) (Slaves: 07) Channel 08: E M (Default) (Slaves: 08) Channel 09: E M (Default) (Slaves: 09) Channel 10: E M (Default) (Slaves: 10) Channel 11: E M (Default) (Slaves: 11) Channel 12: E M (Default) (Slaves: 12) Channel 13: E M (Default) (Slaves: 13) Channel 14: E M (Default) (Slaves: 14) Channel 15: E M (Default) (Slaves: 15) Channel 16: E M (Default) (Slaves: 16) Channel 17: E M (Default) (Slaves: 17) Channel 18: E M (Default) (Slaves: 18) Channel 19: E M (Default) (Slaves: 19) Channel 20: E M (Default) (Slaves: 20) Channel 21: E M (Default) (Slaves: 21) Channel 22: E M (Default) (Slaves: 22) Channel 23: E M (Default) (Slaves: 23) Channel 24: E M (Default) (Slaves: 24) 24 channels configured. Ronald Hartmann Director Technical Services VerCom Systems, Inc. 410 Fame Rd, Dayton, OH 45449 Voice:866.VerCom.4 Fax: 866.422.6486 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avaya firmware
Hi, Brain, I bought an Avaya SIP phone and settup the avaya tftp server and http server. After the first boot, 46xxupgrade.scr and 323tosip1_1.bin were loaded to the phone. But after I rebooted again using DHCP, it keeps rebooting, every time it gets the sip_.txt file from my HTTP server. In sip_.txt file, I configured the firmware download info as the follwos and the two *.ebin files are at the HTTP root directory. BootName sip_4602bt1_1.ebin AppName sip_4602ap1_1.ebin Could you please help me out? I am wondering if avaya 4602 sip phone can be registered with asterisk. I appreciate it! --yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse down?
Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream-Budge tone
Hi; Have two grandstream Budge toneConnected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart I am able to hear voice only if I pressed the hold button and take the call again.This problem also Occurs in calls from x-lite to cisco7940 Does anybody has any idea or documentation about the problem above? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mmkn olmayabilir. Mesaji alan kisi, mesajin gnderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkrler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freeworlddialup
On Wed, 11 May 2005 09:42:07 +0200, Sander wrote: Hi there i just setup my asterisk to dial with freeworlddialup and i am trying to dial 411 voice xml service from freeworlddialup and I always get congestion/busy Is this normal Also dialing the tell number hangs up on me Yeah, I'm seeing the same thing. Dialing 613 (echo test) does the some thing. Funny enough if I dial my own account it does ring through. FWIW, I'm using their IAX connection. Michael FWD 54245 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream-Budge tone
This is usualy a problem with either incompatible codec settings or sometimes when the two systems are not on the same subnet. However, posting every 10 minutes until someone responds is not a good way to get an answer. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül GözlükogluSent: Wednesday, May 11, 2005 6:58 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Grandstream-Budge tone Hi; Have two grandstream Budge tone Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart I am able to hear voice only if I pressed the hold button and take the call again .This problem also Occurs in calls from x-lite to cisco7940 Does anybody has any idea or documentation about the problem above? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse down?
Trevor Harrison wrote: Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor Nope, working fine here ( Modesto California ). Try reversing which gateway you are using first. I did that a while ago and things seem to work fine now. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk not starting
Hi. Well, i guess as the warning says, for some reason the module for ILBC codec cannot be loaded. Try uncommenting ;noload = format_ilbc.so in modules.conf just as a test. If you can start asterisk, then you can be sure thats the problem. Im not sure how to compile again just a module of asterisk, you can do some research about it, or may be some nice guy in this list will show us the light :-) Or, you can recompile All the asterisk system. Best Regards. - moy On 5/11/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi! I have problems with starting asterisk by command asterisk -c My OS is Mandrake 10.1 official and I Am using CAPI with AVM Fritz card and installing it was successful. I have attached lines after asterisk -c with all it's warnings first and also my modules.conf file. I'm pleased with any suggestions you got Thanks in advance! [EMAIL PROTECTED] asterisk]# asterisk -vc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.1-2mdk, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found -- This box has 1 capi controller(s). -- CAPI[contr1] supports DTMF -- CAPI[contr1] supports supplementary services HOLD/RETRIEVE TERMINAL PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR == ast_capi_pvt(50,*,demo,0,2) (1,2,64) == ast_capi_pvt(50,*,demo,0,2) (1,2,64) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD) [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' --
[Asterisk-Users] wip 5000 and using write msg on the phone - anyone?
I have a couple WIP 5000 phones, They are working great. I can use sendtext() and send message to the phone. However, when I select Messages menu on the phone, then write message, I enter a phone extension, some short message and click send. I get FAILED on the screen. I dont see anything on the asterisk console that the phone even tried to connect and send a message or anything. Has anyone tried to use the write msg feature of the phone. I was hoping there would at least have been some error message on the asterisk console but there was not. Thanks for any suggestions. Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] forum www.asterisk-italia.it
Paolo Losi wrote: Hi all! We just opened a forum dedicated to italian asterisk users. It's an attempt to offload some traffic on this huge traffic mailing list and to let italian users to help each other and share experiences. For all italian speaking users please visit and contribute to www.asterisk-italia.it! I don't seem to be able to resolve that link. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject
Thank you and sorryThere is something going wrong with the system I only sent one mail From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 11, 2005 5:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject This is usualy a problem with either incompatible codec settings or sometimes when the two systems are not on the same subnet. However, posting every 10 minutes until someone responds is not a good way to get an answer. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betl Gzlkoglu Sent: Wednesday, May 11, 2005 6:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream-Budge tone Hi; Have two grandstream Budge toneConnected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart I am able to hear voice only if I pressed the hold button and take the call again.This problem also Occurs in calls from x-lite to cisco7940 Does anybody has any idea or documentation about the problem above? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mmkn olmayabilir. Mesaji alan kisi, mesajin gnderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkrler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing calls log in a text file
Greeting!, I read somewhere that without cdr, Mysql etc it is possible to take outgoing-call-logs to a text file. (I am not sure please). is it really possible ? if so, how do I do it? any links to refer? Thank you. Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
I quite like the idea that came about earlier with regards to Romand and Greek gods, I am thinking (if I ever get off the phone to google today) of findind the roman and greek gods of communication.. On 5/11/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On May 11, 2005 03:38 am, David John Walsh wrote: I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Nonsense; it's little sidetracks like this that make the list interesting. You've done absolutely nothing wrong, and I for one am enjoying the different naming schemes (some old, some new) that are coming up here. Don't let the odd social miscreant scare you off. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicepulse down?
it works here in Chicago. you might want to check with your provider their dns may be out.. that happened with comcast about 3 weeks ago. Original Message Subject: [Asterisk-Users] Voicepulse down? From: Trevor Harrison [EMAIL PROTECTED] Date: Wed, May 11, 2005 8:57 am To: asterisk-users@lists.digium.com Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse down?
Its working for me now also Actually, I did try from 2 different ISP's on two sides of the country with the same results. -Trevor On 5/11/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote: it works here in Chicago. you might want to check with your provider their dns may be out.. that happened with comcast about 3 weeks ago. Original Message Subject: [Asterisk-Users] Voicepulse down? From: Trevor Harrison [EMAIL PROTECTED] Date: Wed, May 11, 2005 8:57 am To: asterisk-users@lists.digium.com Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] broadvoice NCFA numbers
call for you. Broadvoice is having issues. For me inbound calling is down (this is day two). Their mantra is : We are currently experiencing in-bound call issues with a carrier partner in some areas. We are aware of the issue and our engineers are working to have it resolved as soon as possible. I've been on hold for over an hour. I'm starting to have doubts that there is anyone answering calls there. Im about to drop their service; looking for another service that allows asterisk and has in bound AZ 480 dids right now. It should say And our ENGINEER not ENGINEERS are hiding under their desk hoping the problem will go away, we highly doubt to have this resolved within a week, please don't call, we don't answer out phones ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
Naming Conventions for Asterisk Hostnames, . Sounds like a Wiki page idea. What is the maximum length of a hostname that will display on all the phones and different displays, do some letters or numbers display baddly on some phones? [EMAIL PROTECTED] is not such a good idea as it is i onley not lonley what is the smallest screen on any phone that could display this name? 20 CHARS? On 5/10/05, David John Walsh [EMAIL PROTECTED] wrote: Hello list we are installing 2 new servers (to run asterisk) shortly, for a stand alone service. Ignoring our current naming convention, we'd like to name them something.. but we are not sure what. a consideration is that on the screens of the phones it shows [EMAIL PROTECTED] (eg [EMAIL PROTECTED]) (all extensions are numeric) so the users will see it everyday i'm not creative in this way, it doesn't need to be a silly reference (like jarjar and anikin etc) per se but im curious what would you name them? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR and Postgres
Hi, I followed the (sort of) instructions on http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+pgsql to enable storing CDR records in a Postgresql database. Actually i only modified the line MODS=cdr_pgsql.so cdr_csv.so #cdr_manager.so for this one: MODS=cdr_pgsql.so cdr_csv.so cdr_manager.so I think have installed the postgres libraries (i mean i have /usr/lib/pgsql.so), im using gentoo and only did emerge postgres, im sure this is enough, because others aplications tha need the pgsql library compile perfectly (i.e. freeradius). Howerver the module is not installed after the make install command, i see the module in voip asterisk # find / -name cdr_pgsql.o /usr/src/asterisk/cdr/cdr_pgsql.o but not in the module directory of asterisk, do i have to copy it manually?, the documentation says it will be copied automatically, what should i do for make this working? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime voicemail login incorrect
I have two ways to go to the voicemail box, either by dialing 8500 from
the phone which received the voicemail (without a password) or from
another phone by dialing 8501 and key in the mailbox and the password.
However, with Realtime the password will be rejected as login incorrect.
What do I miss?
exten = 8500,1,VoicemailMain(s${CALLERIDNUM})
exten = 8500,2,hangup
exten = 8501,1,VoicemailMain
exten = 8501,2,hangup
uniqueid 1
customer_id 602
context other
mailbox 602
password
fullname Ronald Wiplinger
email [EMAIL PROTECTED]
pager
stamp 20050511232451
attach yes
saycid yes
hidefromdir no
*CLI show version
Asterisk CVS-HEAD-04/17/05-07:25:45 built by [EMAIL PROTECTED] on a x86_64
running Linux
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Re: [Asterisk-Users] forum www.asterisk-italia.it
--- Paolo Losi [EMAIL PROTECTED] wrote: For all italian speaking users please visit and contribute to www.asterisk-italia.it! hi, www.asterisk-italia.it could not be found. I only found www.asteriskpbx.it... domé... __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] livevoip
What does this mean? Your account is not working? I made all my LD calls with livevoip today. Yes, my account and that of 16 other users just went down for over 3 days. And what I really resent most of it is the fact that I begged them repeatedly by email to transfer me to another node. They didn't care. I looked bad to my customers, one of them, a new one withdrew, another one took it as an excuse not to pay his due balance. My 2 backup LDs did not responded well because they were SIP based and I had not tested that on my * box in short, I went through unnecessary hell over the weekend, just because you know who did not have the decency to accomodate me. To this day they still do not return my calls. CS -Original Message- From: Tom [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 10, 2005 6:28 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] livevoip At 11:49 PM 5/9/2005, you wrote: They have been down for 3 days now. What does this mean? Your account is not working? I made all my LD calls with livevoip today. Tom Althought I prize their concept, perhaps I am wrong in thinking they are close to the only option. Will anyone know of another carrier that can provide me with (for about 500,000 minsxmonth so far): 1. on the fly dids for $1 2. IAX 3. on the fly 800 setup 4. Level(3) -- Level(3) -- Level(3), I can't stress enough how important a consistent carrier is 5. all incoming/outgoind on the 1.2 range 6. Excellent international long dist rates and quality Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kellner, Peter Sent: Monday, May 09, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] livevoip Yes. I have never had any downtime that I know of but the quality is consistently below average. Because they are iax, it is easy to set up. If you are interested, I'd buy there minimum ($30 I think) and test it out and see what you think. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin Sent: Monday, May 09, 2005 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] livevoip Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 driver compiling problem.
i use asterisk cvs head ( two days ago) more or less openh323 1.12.2 (oh323 home page) and pwlib 1.5.2 (oh323 home page) asterisk-oh323-0.7.2-pre1 library versions? where download? versions from oh323 readme are not in sourceforge site. but i obtain this error compiling: [EMAIL PROTECTED]:/usr/src/asterisk/cvs/last/asterisk-oh323-0.7.2-pre1# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/cvs/last/asterisk-oh323-0.7.2-pre1/wrapper' ./check_ver /usr/src/asterisk/cvs/last/pwlib pwlib ./check_ver /usr/src/asterisk/cvs/last/openh323 openh323 g++ -DP_LINUX=2.4.30-rc3 -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/usr/src/asterisk/cvs/last/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/usr/src/asterisk/cvs/last/pwlib/include -DPTRACING -I/usr/src/asterisk/cvs/last/openh323/include -DHAS_IXJ -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.5.2\ -DOPENH323VERSION=\1.12.2\ -DP_HAS_IPV6 -I/usr/src/asterisk/cvs/last/pwlib/include -I/usr/src/asterisk/cvs/last/openh323/include -I/usr/src/asterisk/cvs/last/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cvs/last/asterisk-oh323-0.7.2-pre1/wrapper' make: *** [subdirs_build] Error 1 -- César García. Director de Sistemas, IdecNet S.A. Centro de Gestión de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - España. Tfn: +34 828 111 000 Ext: 340 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live Voip
Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is it allowed to install 2 TE405P cards at same P.C.?
On Wed, 11 May 2005 07:31:17 +0300, Yousri Farouk wrote Hello Does Asterisk allow to install two pci TE405P Cards at the same P.C.? You should be able to do it as the limit for Zap channels is 255. I do not know if the computer can handle the interrupt load generated by two cards. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse down?
gwiaxt01.voicepulse.com working fine 5 mins ago, just used it. Mike On 5/11/05, Trevor Harrison [EMAIL PROTECTED] wrote: Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote: On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. There's a whole french comic suited to an Asterisk naming convention. I'll leave it as an exercise for the reader ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 Skype/In callto://stevekennedyuk / UK callto://+442088167166 US callto://+13106518226mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On the surface, naming may sound trivial. When you're dealing with users and people paying your salary/consulting fees, it's not. Offending the wrong person because you named the server Nag, Chatterbox, ETPhoneHome, etc. can be very costly. You could use some of the following: PhoneSystem PhoneManager PhoneVoiceMail CompanyPhone Or, if the atmosphere is right, use something cute: Lt.Uhura RadarOriely Ken Gents Whilst I apreciate the sentiments regarding my question, if you are to look at my track record of helping people - across in the majority [EMAIL PROTECTED], AMP lists and to a lesser (but growing) extent asterisk-user and asterisk-biz, its not up there with the super gurus, but I am putting more back into the list. I have asked my fair share of questions as well, (and will continue to do so when needed). I noticed that you didn't make these comments when people have talked exclusivly about polycom or cisco hardware (as technically these are not asterisk either) There is a high technical aim to the list, there is also a community factor. I would however kindly ask that for the benifit of the list any further discussions regarding this are directed to me personally, the header information contains my email adress. Regards David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live Voip
They're really slow to set things up. I signed up 2 nights ago.. still waiting. I'm going to give the the benefit of the doubt.. but am signing up for other voip services in the mean time. JD My question to them: It's been over 24 hours. I figured out how to login, my accountcode appears to be X. The DID I selected is not attached to the account as far as I can tell. I would think this sort of thing would be more automated. JD Their response: We are not automated. The reason is security. Even credit card charges are all verified. Its all to protect the system from abuse. If we did not verify things the amount of people who would attempt fraud would be un-real. Our techs are just working alot of orders and do everyone in the order they arrived. Sales LiveVoip LLC Sean Kennedy wrote: Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What do you name yours
Personally, I always liked TuxPBX.SomeDOmain.com As for Greek and Roman Gods of communication... Mercury for the Romans... Hermes for the greeks. He was Zeus' messenger For large growth systems, country names are very popular. Otherwise, simple names of whatever fictional group makes yo happy is cool. X-men characters for example or maybe Star Wars planets or whatever works Have fun, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David John Walsh Sent: Wednesday, May 11, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] What do you name yours I quite like the idea that came about earlier with regards to Romand and Greek gods, I am thinking (if I ever get off the phone to google today) of findind the roman and greek gods of communication.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
trixter http://www.0xdecafbad.com wrote: On Tue, 2005-05-10 at 23:46 -0400, Black Ratchet wrote: My two toy boxes at work are 'succasunna', named after the 1st city to ever have #1ESS phone switch, and 'murrayhill', which was named after the location of the ATT headquarters. I wonder what that big building in basking ridge was then. Verizon HQ? ;-) To those that don't understand, it looks like Verizon has purchased the Basking Ridge building. I think it was initially sold to an insurance company but I'm not sure. And Murray Hill was the Bell Labs headquarters, I think. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Avaya firmware
Yes, Avaya 4602 works with asterisk. I followed usual firmware change procedure and successfully changed firmware to SIP. I was not using DHCP. To change software - there is no need to change txt files supplied with software package - at least defaults worked for me. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Yao, YuanbinSent: Wednesday, May 11, 2005 4:53 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Avaya firmware Hi, Brain, I bought an Avaya SIP phone and settup the avaya tftp server and http server. After the first boot, 46xxupgrade.scr and 323tosip1_1.bin were loaded to the phone. But after I rebooted again using DHCP, it keeps rebooting, every time it gets the sip_.txt file from my HTTP server. In sip_.txt file, I configured the firmware download info as the follwos and the two *.ebin files are at the HTTP root directory.BootName sip_4602bt1_1.ebinAppName sip_4602ap1_1.ebinCould you please help me out? I am wondering if avaya 4602 sip phone can be registered with asterisk.I appreciate it!--yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail login incorrect
Where is your debug trace?
-Matthew
Ronald Wiplinger wrote:
I have two ways to go to the voicemail box, either by dialing 8500
from the phone which received the voicemail (without a password) or
from another phone by dialing 8501 and key in the mailbox and the
password. However, with Realtime the password will be rejected as
login incorrect. What do I miss?
exten = 8500,1,VoicemailMain(s${CALLERIDNUM})
exten = 8500,2,hangup
exten = 8501,1,VoicemailMain
exten = 8501,2,hangup
uniqueid 1
customer_id 602
context other
mailbox 602
password
fullname Ronald Wiplinger
email [EMAIL PROTECTED]
pager
stamp 20050511232451
attach yes
saycid yes
hidefromdir no
*CLI show version
Asterisk CVS-HEAD-04/17/05-07:25:45 built by [EMAIL PROTECTED] on a x86_64
running Linux
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Re: [Asterisk-Users] is it allowed to install 2 TE405P cards at same P.C.?
On May 11, 2005 11:45 am, Carlos Chavez wrote: You should be able to do it as the limit for Zap channels is 255. I do not know if the computer can handle the interrupt load generated by two cards. AFAIK there is no zap channel limit; yes there are 255 device nodes but I think everything just uses /dev/zap/ctl and opens the channels that way. Also, the interrupt load is no worse for two TE405Ps as it is for two X101Ps; 1000 ints/sec/card. I don't know enough about the driver design to know how much more work is actually done per interrupt, though. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T-1 (NOT PRI) Voice and IP mixed
Hello All, I have googled and wikied but must not be searching correctly. Assuming the TE110P has same ability as old T100P to use some voice and some data channels, lets say I have a TE110P set to accept voice on 10 channels and pass the other 14 channels as data. Under this scenerio i am guessing that * should still be able to accept VoIP calls on the data channels and still allow internal user to access Internet through data portion of the T? Am I correct or am I talking out of my ... I know that a channel bank would be a better solution, but. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial * get dial tone, push pin dial out......
I need some help...using @home v1.0 with TDM11B and 4 trunks..everything working fine internally I need to be able to dial into * , get dial tone and pust a code (PIN) or whatever 4 digits and be able to call out, what I need?? Samples exten.conf please? THANKS Manny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On Wed, 2005-05-11 at 17:11 +0100, Steve Kennedy wrote: There's a whole french comic suited to an Asterisk naming convention. I'll leave it as an exercise for the reader ... Yes, before I got involved with Asterisk I had machines at customers called Asterix, Obelix etc. but watch out remember Mobilix had problems with the publishers. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI QSIG and legacy toshiba intergration
I love it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTNCTX670Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx toa remote ctx100 (this is setup using QSIG ISDN on a PRI tie line) now I would like to have 8xxx going to asterisk All of my incoming calls would be handled by the ctx670, mostly on DNIS equipted lines. If the user dials a four digit extension starting with "8" on the CTX 670, how do I transfer that digit string to asterisk? Going backwards, the asterisk user dialing a CTX extension, I plan to handle this with DID digits sent back to the CTX, when I see the Incoming DID digits I can route the call based on DID I receive from asterisk on that particular line group. It will sort work like this... Ext.123 is dialed on asterisk Asterisk picks up a zap channel sends DTMF 123 after the CTX picks up CTX looks at DID and sends it to the destination 123 I think that I need a context with all of my CTX extensions in it! Ultimately I am trying to find out if Qsig can somehow help me, I am more of a phone guy, I need an asterisk guy to exchange help with.. This e-mail message and any attachments may contain information that is privileged and confidential. The information contained in this e-mail is intended only for the use of the addressee; access by anyone else is unauthorized. If this message has been sent to you in error, do not review, disseminate, distribute or copy it. Please immediately reply to the sender by e-mail or by telephone at608-783-7560 X299,then delete the message and any attachments from your system. Thank you for your cooperation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . We're using sleepy, dopey, grumpy, doc, etc. for all the machine. Assigned appropiately, doc is the monitoring system, grumpy is the one remining Win2k server, etc. -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]522 Black Canyon Park p:404-932-1355 f:866-751-6494 Canton, GA 30114 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound Calls Codec
I'm noticing by watching the CLI that my inbound calls coming via T1s on a TE410P are using GSM codec. Why wouldn't it use ULAW as default? How can I make it use ULAW as default? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mass Deployment
With the lack of information on Mass Deployment or deployment in general I have been working on both some wiki pages and some software. The software is PHP based and writes the TFTP files or other files needed to boot and run Ciscos, Snoms, and what ever it might take. What is the interest in such software? It has some securtiy issues with what it can do so I am holding off release at this time. What other phones/devices should it support? Who can get me the documentation? Should I interface it with Asterisk Realtime or should I keep it clean and seperate? This tool, correctly installed could allow remote installation of hundreds of phones. Do people really need hand holding or onsite consulting? I am open to all ideas, keep it on the list so good ideas are not lost. -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] broadvoice NCFA numbers
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Dziubanski Sent: Wednesday, May 11, 2005 7:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] broadvoice NCFA numbers Im about to drop their service; looking for another service that allows asterisk and has in bound AZ 480 dids right now. It should say And our ENGINEER not ENGINEERS are hiding under their desk hoping the problem will go away, we highly doubt to have this resolved within a week, please don't call, we don't answer out phones In all fairness, they do answer their phones, and (in rather stark contrast to some other VOIP providers) their support staff are remarkably pleasant under what must be extremely trying circumstances. The wait is long, to be sure, but there are humans there. The human I spoke to took time to do what trouble-shooting he could, was grateful to have a clue-ful asterisk user that could at least tell him what errors were being returned, and then eventually explained that the problem was with a vendor and that the ETR supplied by the vendor was long passed. He could offer no realistic estimate for restoration of service. It's a shame that they can't get things fixed. This episode is going to cost them dearly, because I believe that lots of others are doing what I'm doing, and pulling the plug. It's been a week now, and there are other options. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 05/10/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble Connecting Xlite to Asterisk
I just installed Xorcom Rapid and I'm trying to connect with Xlite. In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address of the new install. I can ping that box. When I try to connect I get hung on the Awaiting Proxy login information and the log reads: © 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1103m build stamp 14262 License key: A27D1192D9FA4B609F02F3AC31B6BD12 Established SIP protocol listen on: 172.16.17.99:5060 Discovered Port Restricted Single Mapped Port Symmetric NAT Firewall SIP: 172.16.17.99:5060 RTP: 172.16.17.99:8000 NAT: 204.94.248.12 Discovering external SIP port on symmetric firewall... The Asterisk server and my machine are on the same subnet. I just want to make a connection for testing purposes before I proceed with the setup. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Passwords
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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Re: [Asterisk-Users] Realtime voicemail login incorrect
Matthew Boehm wrote:
Where is your debug trace?
Have you had a look at my settings already? Do you think it is correct?
Or have you seen a mistake?
If yes, yes and no, than please guide me to a page how I can produce the
debug trace.
I consider to DEBUG as the last methode to solve something, and than I
suspect a bug. I don't suspect a bug, but a mistake in my settings.
bye
Ronald
-Matthew
Ronald Wiplinger wrote:
I have two ways to go to the voicemail box, either by dialing 8500
from the phone which received the voicemail (without a password) or
from another phone by dialing 8501 and key in the mailbox and the
password. However, with Realtime the password will be rejected as
login incorrect. What do I miss?
exten = 8500,1,VoicemailMain(s${CALLERIDNUM})
exten = 8500,2,hangup
exten = 8501,1,VoicemailMain
exten = 8501,2,hangup
uniqueid 1
customer_id 602
context other
mailbox 602
password
fullname Ronald Wiplinger
email [EMAIL PROTECTED]
pager
stamp 20050511232451
attach yes
saycid yes
hidefromdir no
*CLI show version
Asterisk CVS-HEAD-04/17/05-07:25:45 built by [EMAIL PROTECTED] on a x86_64
running Linux
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Re: [Asterisk-Users] good bri card not junghanns
What is the problem with junghanns? I use them without problems. klaus Eugenio De Vena wrote: Hi there, will someone suggest me a good and * combatible isdn card ( 1 , 2 , 4 , 8 channels ). I am currently working with but can not stand their complete lack of support. I have tried DIVA 4-M and it seems good but 1) do not know about stability 2) drivers are from junghanns so the problem of support remains. Any hint? Thanks in advance Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Predictive Dialers
I took a look but was wondering if there are any other options out there? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Nuno Viegas |Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Predictive Dialers | |Hi Anton, | |Start by having a look at this: | | http://www.voip-info.org/wiki-Predictive+dialer | |N | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: 11 May 2005 10:19 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Predictive Dialers | |Guys. | |Anybody know of any predictive dialers for Asterisk and Windows? | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transparently Routing German pri through Asterisk
Hi!
Be also aware of the ISDN TON (type of number: national, international,
...). Asterisk stable does not handle them correct - there are some
patches in CVS or in bristuff. If your PBX or the telco uses varying
contexts (typical for the CLI), you might experience problems with the
CLI on the phones (missing 0 in front of the number ...).
Also you might have problems with G4 fax which use the high information
layer.
regards,
klaus
Jay Milk wrote:
Sure... except that the second call *would* be to extension h and not
to the same number. Haven't observed that behavior in real life,
though. Have you?
Secondly, in my setup, we're using the proper 10 and 7-digit and 011X
patterns. However, this was a suggestion for a German PRI, and all bets
are off for that, as German numbers don't follow very specific rules.
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Saturday, May 07, 2005 4:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transparently Routing German
pri through Asterisk
Jay Milk wrote:
You got your groups mixed up. Should be:
[default]
exten = _.,1,Dial(ZAP/g2/${EXTEN})
[outgoing]
exten = _.,1,Dial(ZAP/g1/${EXTEN})
Means that anything coming in to channel-group 1 (default context)
will be sent out through group 2, and vice versa.
Watch the console and be amazed when _. matches extension
h, which is
called when the far side of the call hangs up. You get two
calls to the
same number by only dialing once! Stop being lazy and at
least use _X.
as your pattern. ___
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Re: [Asterisk-Users] Group dial, first phone cannot pickup call if included in subsequent steps.
I've tried it on another system and can reproduce the fault at will. It would seem that the first Dial command is not terminated when the second Dial includes the first extension. e.g. exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1002,10,tr) exten = 100,3,Dial(SIP/1003,10,tr) Will do exactly what you'd expect, dials exten 1001 for 10 secs, then 1002 for 10 secs, and finally 1003 for 10 secs. However once you add extra extensions it all goes horribly wrong. e.g. exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1002SIP/1003,10,tr) This does what is expected, after ringing extension 1001 for 10 secs extensions 1002 1003 start ringing and the first one to answer gets the call where upon the second one stops ringing. However the following is nastier: exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1001SIP/1003,10,tr) After ringing extension 1001 for 10 secs extensions 1001 and 1003 start ringing but only extension 1003 can pickup the call. If extension 1001 is picked up after extension 1003 has started ringing it is dead as a dodo. Is a bug or a feature? On Tue, 2005-05-10 at 09:46, bam wrote: I have a simple dial plan to cascade calls when the first phone does not answer: exten = 100,1,Dial(SIP/1000,10,tr) exten = 100,2,Dial(SIP/1000SIP/1001,10,tr) exten = 100,3,Dial(SIP/1000SIP/1001SIP/1002,10,tr) exten = 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user picks up the phone it is just dead, no dial tone, nothing. Occasionally the handset will hang and need to be power-cycled. I've swapped out the phone, the power supply, and even the cabling, but no joy. As long as exten 1000 picks up the call at step one everything works fine. Apart from that everything else seems tickety-boo. Cisco 7905s with latest SIP build and Asterisk CVS-HEAD-03/02/05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Log Output
Thx Julian |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Julian J. M. |Sent: Miércoles, 11 de Mayo de 2005 03:46 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Log Output | |In /etc/asterisk/logger.conf, add this: | |full = notice,warning,error,debug,verbose | |Then watch /var/log/asterisk/full getting really big ;) | |Julian. | |On 5/11/05, Anton Krall [EMAIL PROTECTED] wrote: | Guys. | | Is there a way to output the same information shown on the console | when invoked as - but to a log file for later |grepping and such? | | I noticed the normal log only shows warning and errors but no info | messages like in the console.. Any ideas? |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI QSIG and legacy toshiba intergration
Jordan, dont beat yourself up, people are buying legacy Cisco equipment today. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Wednesday, 11 May 2005 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PRI QSIG and legacy toshiba intergration I love it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTNCTX670Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx toa remote ctx100 (this is setup using QSIG ISDN on a PRI tie line) now I would like to have 8xxx going to asterisk All of my incoming calls would be handled by the ctx670, mostly on DNIS equipted lines. If the user dials a four digit extension starting with 8 on the CTX 670, how do I transfer that digit string to asterisk? Going backwards, the asterisk user dialing a CTX extension, I plan to handle this with DID digits sent back to the CTX, when I see the Incoming DID digits I can route the call based on DID I receive from asterisk on that particular line group. It will sort work like this... Ext.123 is dialed on asterisk Asterisk picks up a zap channel sends DTMF 123 after the CTX picks up CTX looks at DID and sends it to the destination 123 I think that I need a context with all of my CTX extensions in it! Ultimately I am trying to find out if Qsig can somehow help me, I am more of a phone guy, I need an asterisk guy to exchange help with.. This e-mail message and any attachments may contain information that is privileged and confidential. The information contained in this e-mail is intended only for the use of the addressee; access by anyone else is unauthorized. If this message has been sent to you in error, do not review, disseminate, distribute or copy it. Please immediately reply to the sender by e-mail or by telephone at608-783-7560 X299,then delete the message and any attachments from your system. Thank you for your cooperation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channelized T-1 (NOT PRI) Voice and IP mixed
John Millican wrote: Hello All, I have googled and wikied but must not be searching correctly. Assuming the TE110P has same ability as old T100P to use some voice and some data channels, lets say I have a TE110P set to accept voice on 10 channels and pass the other 14 channels as data. Under this scenerio i am guessing that * should still be able to accept VoIP calls on the data channels and still allow internal user to access Internet through data portion of the T? Am I correct or am I talking out of my ... I know that a channel bank would be a better solution, but. John M You'll need to have hdlc enabled in your kernel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: snom mass deployment - settings via DHCP
I have to adjust my last statement. If Setting URL field of advanced.htm webinterface is empty the value of tftp-server-name is used. option tftp-server-name http://192.168.100.1;; On reboot the phone sends two requests to the specified IP: GET /download/snom190.htm GET /download/snom190-mac.htm bootfile-name does indeed seem to be ineffective. I tried using snom 190 with 3.60b firmware and a dhcp-3.0.1 server. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Satellite Providers
Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live Voip
I personal would not touch LiveVoIP with a 10 foot pole. They don't provide very little customer service. The service they do offer has been very bad or unusable in the past. On Wed, 2005-05-11 at 10:53, Sean Kennedy wrote: Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Passwords
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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Re: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmware upgrade
I saw such wired behaviours on several SPAs. I think this is caused defect flash. We sent our SPAs back and got a new HW Version. regards klaus Steve Prior wrote: I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. Thanks Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio delays during file playback and zap channel activity
Hi - I've noticed that I'm getting audio delays when asterisk is playing back a file from disk and new zap channels are being created or destroyed. Audio streams are generally fine (person to person calls do not experience this issue). Sometimes the drops are very short - barely noticeable. Sometimes they are up to 1 second, and whatever file is being played resumes at the same spot after the drop is done (E.G. On a voicemail call, Allison says You have five old messages. Pr [PAUSE] ess one for). The issue exists across all file types (gsm, wav, mp3, etc). It seems to happen on every zap channel creation/destruction, but some are so short that it just sounds like an audio anomaly rather than a delay. When there are no active zap channels, the issue does not exist. The disk controller and the Zap card do not share an interrupt (The zap card has its own interrupt - not shared with anything). We're running a version of CVS HEAD from 04/09/05 on a Dell PE 1600SC with a U320 SCSI, Raid 1 (LSI Logic / Symbios Logic 53c1030 PCI-X Fusion-MPT Dual Ultra320 SCSI), with a TE410P connected to a PRI. Has anybody experienced anything similar? Anybody have a clue as to what's going on? Zttool does show 15 IRQ misses, but that is over the course of the past week, and after new incidents the number of misses stays the same. Thanks, Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RED ALARM on PRI channel takes Asterisk DOWN (FIXED)
Sangoma is great prompt response to the problem, its fixed now Use the latest stable latest stable 2.3.2-3 +++ Vikram Rangnekar [05/05/05 08:58 +0200]: have an asterisk box (P4, 1GB RAM, etc ) with a sangoma two E1 port card in it. I have used a E1 cross cable to connect the two E1 ports together and pass calls through them for testing purposes allo works fine. also have multiple voip phones connected to this setup, what i noticed is that when i pull any one end of the E1 (breaking the E1 connection) I get multiple RED ALARMS on the zap channels I understand this is ok and should happen if the E1 link breaks but my problem is that asterisk stops doing a lot of other things too like i have extensions configured to meetme, voicemail and other aplications they all stop working i get no sound but when i call one hardphone to another it works i guess thats cause the rtp streams go from phone to phone directly. but why does asterisk stop passing anything through itself. Also if anyone else has experience this is this only with sangoma or does it happen with asterisk too. I have a digium card but its a single port and i dont have a real E1 link to test with. ) Few lines from the dump i get on the asterisk cli. the first few lines are the red alarm and the rest of me trying to call up an extension -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom190 and SUBSCRIBE failures with 407
Hi, I have searched and searched, but cannot identify what is happening here... I have several snom190 phones, and all of them have the 5th function key set to call asterisk by using the destination option. This automatically causes the phone to SUBSCRIBE for NOTIFY messages for the asterisk destimation. Asterisk does not seem to like the SUBSCRIBE requests. It logs the message: NOTICE[22527]: chan_sip.c:7601 handle_request: Failed to authenticate user sip:[EMAIL PROTECTED];tag=blw952f4zl for SUBSCRIBE Or similar for every subscribe... The phone is registered happily, and can make and receive calls. I have set up a hint for the s extension, although I do not think this should be necessary. I checked the SIP packets, and Asterisk is responding with SIP Status: 407 Proxy Authentication Required, but if I look at the request, all of the authentication data was provided (in md5 format) in the original SUBSCRIBE request. Any clues out there? This is causing really annoying messages on the snom phone's screen as it thinks it needs a password entering. Thanks. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with MeetMe
I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm getting the following problem: -- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack == Parsing '/etc/asterisk/meetme.conf': Found May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such device or address May 11 14:05:46 ERROR[20050]: chan_zap.c:6687 chandup: Unable to dup channel: No such device or address May 11 14:05:46 WARNING[20050]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device May 11 14:05:46 WARNING[20050]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I have the following in meetme.conf [rooms] conf = 0224 What could be happening? I don't have any digium cards on the machine. lsmod shows: Module Size Used byNot tainted zaptel182080 0 The other modules are not related to zaptel or asterisk. ls -l /dev/zap/ps* shows: crw-r--r--1 root root 196, 255 May 11 10:07 /dev/zap/pseudo Any ideas? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX and calls on hold
Hello - I recently offloaded some of the SIP traffic on to a seperate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction. Basically the call comes in via a ZAP channel over a PRI into our main system, is fed over IAX to our second system and then is connected to the SIP channel (client). I've tried both enabling and disabling IAX trunking and jitterbuffers. I've also added a zap card and enabled it to allow for a timing source. The new system is running the latest CVS of Asterisk and libraries as of yesterday, while the other one is running a CVS version as of Jun of last year. I'm using RSA for auth between the servers (IAX). Any help would be appreciated. Thanks. Jeroen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Predictive Dialers
What exactly are you looking for? There are basically 3 commercial solutions: Aheeva, DACX and Sinedialer and there are 2 open-source solutions: ShadyDial and VICIDIAL What features do you need that are not addressed by one of these? MATT--- -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 11, 2005 1:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Predictive Dialers I took a look but was wondering if there are any other options out there? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Nuno Viegas |Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Predictive Dialers | |Hi Anton, | |Start by having a look at this: | | http://www.voip-info.org/wiki-Predictive+dialer | |N | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: 11 May 2005 10:19 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Predictive Dialers | |Guys. | |Anybody know of any predictive dialers for Asterisk and Windows? | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
very nice touch! I like that - apart from parc asterisk (the theme park just outside Paris) I went there as child and went on the seven loop roller coaster, as we went around the loop, we saw something drop past (i thought it was someones glasses / wallet) it was a wheel. still went on it again an hour later once they put it back on!!! David (it was asterisk related, and I was a user of their service!) On 5/11/05, Steve Kennedy [EMAIL PROTECTED] wrote: On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote: On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. There's a whole french comic suited to an Asterisk naming convention. I'll leave it as an exercise for the reader ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 Skype/In callto://stevekennedyuk / UK callto://+442088167166 US callto://+13106518226mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
