[Asterisk-Users] voicemail not working with mysql !!!
I am a newbie to asterisk if u please sort this out... and kindly tell me how to mail to ur mailing lists...whose archives are on www.mark.net I need some tips on configuration of voicemail with mysql... here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root format=wav serveremail=asterisk attach=yes maxmessage=60 maxgreet=60 maxlogins=3 [default] 1234 => 7654,Atif Rasheed,[EMAIL PROTECTED] **voicemail.conf*** I have created the database"asteriskvmusers" in mysql and then created the table 'users' in that database. mysql> select * from users; +-+-+--+--++---+-++ | context | mailbox | password | fullname | email | pager | options | stamp | +-+-+--+--++---+-++ | default | 1234| 7654 | Atif Rasheed | [EMAIL PROTECTED] | | | 00 | +-+-+--+--++---+-++ but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... one more thing which one is newer version, and has mysql support... voicemail or voicemail2 please figure this out... Thank you -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] on hold music from a mp3 stream or sound card input?
On Thu, 4 Mar 2004, Dean Collins wrote: > > How large can the mp3 file be? I haven't played with this yet but > wondering if I can connect about 20-30 mp3's together so my people on > hold don't hear the same music very often. > You can specify a directory full of mp3's that * will pass to mpg123 to play in random order. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] on hold music from a mp3 stream or sound card input?
No need to string them together. Just put them in the MP3 directory and it will play them one by one, taht's all i have done. My largest MP3 plays for 20 minutes. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dean Collins Sent: 04 March 2004 07:05 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] on hold music from a mp3 stream or sound card input? Can I ask an addendum question to this. How large can the mp3 file be? I haven't played with this yet but wondering if I can connect about 20-30 mp3's together so my people on hold don't hear the same music very often. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, 4 March 2004 12:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] on hold music from a mp3 stream or sound card input? Hi Folks, Rather than have my hold music play from a sound file I'd like to have a live feed from a sound card input or MP3 stream. Is this doable and if so how? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Palm OS5 client
Does anyone know of a Palm OS5 client that can connect to asterisk? Hopefully I can use gprs to connect back to my home pabx and make local calls while on the road. Also can anyone comment on how well the CE clients work? Cheers, Dean
RE: [Asterisk-Users] on hold music from a mp3 stream or sound card input?
Can I ask an addendum question to this. How large can the mp3 file be? I haven't played with this yet but wondering if I can connect about 20-30 mp3's together so my people on hold don't hear the same music very often. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, 4 March 2004 12:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] on hold music from a mp3 stream or sound card input? Hi Folks, Rather than have my hold music play from a sound file I'd like to have a live feed from a sound card input or MP3 stream. Is this doable and if so how? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best ATA 186 Firmware
ATA 186 is Cisco, not grandstream From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug HarrisSent: Thursday, March 04, 2004 12:52 AMTo: Asterisk UsersSubject: Re: [Asterisk-Users] Best ATA 186 Firmware Where can I download this version ? Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/ Thanks Doug Message: 10 Date: Wed, 03 Mar 2004 23:50:40 +0200 From: NetOne Administrator <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best ATA 186 Firmware Reply-To: [EMAIL PROTECTED] v.3.0 works fine too James Coberly wrote: > v2.16.2 ata18x > >Works Fine for me. > >
[Asterisk-Users] Handling of AddQueueMember error
I'm trying to put together the right entries in extensions.conf for agents to logon to a queue. It works, but I want to handle the case where an agent logs on a second time. I want to Playback(agent-loginok), but I don't think that * returns priority n+101. extensions.conf ; ; Support queue logon ; exten => 998,1,AddQueueMember(helpdeskq) exten => 998,2,Playback(agent-loginok) exten => 998,3,Hangup exten => 998,102,Playback(agent-alreadyon) exten => 998,103,Hangup Here's what I get at the console: Mar 2 23:10:52 WARNING[688146]: app_queue.c:1073 aqm_exec: Unable to add interface 'Zap/2' to queue 'helpdeskq': Already there == Spawn extension (internal, 998, 1) exited non-zero on 'Zap/2-1' I looked at the source code (apps/app_queue.c) and the only thing happening there is a call to ast_log() when interface_exists != NULL. Thanks, Francois ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 hardware problems?
John Morris wrote: Hi! I didn't get an answer on this, so I'm going to be annoying and try again. For some reason, it looks like two of my channels on my TDM400 stopped working for no good reason. Asterisk stopped working on my main extension today (it does this every week or so). I usually remedy this by stopping asterisk, removing the kernel modules, reinstall them, and restart asterisk, everything's good again. I did this today, but at the step where I install the modules, the kernel hung, with this in /var/log/messages: Have you tried cold-starting your machine? When we first got the TDM boards, I seem to remember some things like this. I shut the machine down, reseated the board, daughter boards, and power supply connector, and it came up clean and has been fine since. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme 'd' and 'p' flags mutually exclusive with wcfxo driver but not ztdummy
Hi List, I'm having some strange behaviour with the MeetMe app. Recently installed a x100p on a friend's box. I've been using conferencing at home fairly extensively, but have no zap interface here, so have been using ztdummy for my timer. When I ported some of my extensions over to his box, I discovered something odd. Dynamic conferencing and the exit on '#' features dont seem to play well together. Moreover, the order of the flags in the argument to the MeetMe app determines the behaviour. That is, if I pass the 'd' flag first, the conference is created, but I cannot exit... It was suggested to me on IRC that the parking could be 'stealing' my # keypress away, but I am not sure this is the case, as the channel remains connected to the MeetMe conference, and further keypresses fail to produce any results either. On the otherhand, if I pass the 'p' flag first, I get an invalid conference error. the 'p' flag works fine on static conferences. Is this a bug, or a possible driver issue, or .. any ideas? I'm still learning on the configuration of zaptel devices and thier modules, but I have the card installed and tested and it seems to work fine. I have only modprobed the wcfxo driver, which loaded that and the zaptel driver, which I gather is the timing interface that meetme needs to run. If anyone thinks this might be system specific behaviour, or you can verify that you have/dont have this problem, please let me know and i'll post my lsmod info and all that good stuff for comparison. Thanks in advance, brook. --- _ Get business advice and resources to improve your work life, from bCentral. http://special.msn.com/bcentral/loudclear.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 hardware problems?
On Thu, 3 Mar 2004, John Morris wrote: >So I reboot, and try to bring things back up again. But now the first >two channels on my TDM400 aren't working, a new one. The light on >them isn't glowing, and installing the module I get this: > >Feb 29 01:09:15 squid kernel: Zapata Telephony Interface Registered on major 19\6 >Feb 29 01:09:18 squid kernel: Freshmaker version: 62 >Feb 29 01:09:28 squid kernel: Freshmaker passed register test >Feb 29 01:09:28 squid kernel: Timeout waiting for calibration of module 0 >Feb 29 01:09:28 squid kernel: ProSlic died on Calibration. >Feb 29 01:09:28 squid kernel: Module 0: Not installed >Feb 29 01:09:28 squid kernel: Timeout waiting for calibration of module 1 >Feb 29 01:09:28 squid kernel: ProSlic died on Calibration. >Feb 29 01:09:28 squid kernel: Module 1: Not installed >Feb 29 01:09:28 squid kernel: Module 2: Initialized >Feb 29 01:09:28 squid kernel: Module 3: Initialized >Feb 29 01:09:28 squid kernel: Found a Wildcard FXS: Wildcard S400P Prototype (4\ >modules) > >Luckily, at least the last two are working, so I plugged my extensions >into those, modified extensions.conf to reflect the change, and I'm >working again for the moment. The archives say Martin suggested something to try for this error: http://lists.digium.com/pipermail/asterisk-users/2003-August/017174.html Have you tried reseating the modules on the card? If you switch modules 0,1 with 2,3 does the problem follow? Other than that, I'm not of much help to you Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 4 Port IADS
I am looking for recommendations on SIP IADs (4 to 8 port) that have been shown to work well with asterisk, I have seen talk about the mediatrix and audiocodes but have seen mixed reviews from the list on interop. Could someone comment or send reviews of their experience with SIP IADs and their functionality (or lack there of) when working with asterisk. All help is appreciated. I will need these asap so if anyone has a couple laying around they can ship quickly feel free to email me off line. Thanks, Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 hardware problems?
Hi! I didn't get an answer on this, so I'm going to be annoying and try again. For some reason, it looks like two of my channels on my TDM400 stopped working for no good reason. Asterisk stopped working on my main extension today (it does this every week or so). I usually remedy this by stopping asterisk, removing the kernel modules, reinstall them, and restart asterisk, everything's good again. I did this today, but at the step where I install the modules, the kernel hung, with this in /var/log/messages: Feb 29 00:51:01 squid kernel: Found a Wildcard FXO: Wildcard X101P Feb 29 00:51:01 squid kernel: Found a Wildcard FXO: Wildcard X101P Feb 29 00:51:01 squid kernel: PCI: Enabling device 00:0c.0 ( -> 0003) Feb 29 00:51:01 squid kernel: Freshmaker version: ff Feb 29 00:51:01 squid kernel: 00 != ff Feb 29 00:51:01 squid kernel: 01 != ff [] Feb 29 00:51:01 squid kernel: 0e != ff Feb 29 00:51:01 squid kernel: 0f != ff So I reboot, and try to bring things back up again. But now the first two channels on my TDM400 aren't working, a new one. The light on them isn't glowing, and installing the module I get this: Feb 29 01:09:15 squid kernel: Zapata Telephony Interface Registered on major 19\6 Feb 29 01:09:18 squid kernel: Freshmaker version: 62 Feb 29 01:09:28 squid kernel: Freshmaker passed register test Feb 29 01:09:28 squid kernel: Timeout waiting for calibration of module 0 Feb 29 01:09:28 squid kernel: ProSlic died on Calibration. Feb 29 01:09:28 squid kernel: Module 0: Not installed Feb 29 01:09:28 squid kernel: Timeout waiting for calibration of module 1 Feb 29 01:09:28 squid kernel: ProSlic died on Calibration. Feb 29 01:09:28 squid kernel: Module 1: Not installed Feb 29 01:09:28 squid kernel: Module 2: Initialized Feb 29 01:09:28 squid kernel: Module 3: Initialized Feb 29 01:09:28 squid kernel: Found a Wildcard FXS: Wildcard S400P Prototype (4\ modules) Luckily, at least the last two are working, so I plugged my extensions into those, modified extensions.conf to reflect the change, and I'm working again for the moment. Anyone see something I'm missing here? Another similar thread on the list, someone said maybe it's a bad module, but I don't understand why *two* would suddenly stop working at the same time after many months. I am running a rather old asterisk version, 0.4.0, and whatever zaptel/zapata modules were current then, with kernel 2.4.20. Thanks for any help! John -- John Morris +1-512-480-0200x1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA 186 Firmware
Where can I download this version ? Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/ Thanks Doug Message: 10 Date: Wed, 03 Mar 2004 23:50:40 +0200 From: NetOne Administrator <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best ATA 186 Firmware Reply-To: [EMAIL PROTECTED] v.3.0 works fine too James Coberly wrote: > v2.16.2 ata18x > >Works Fine for me. > >
[Asterisk-Users] More recordings for Allison
I've been asked to do another sound recording run for words for Allison. For the current more-or-less list of words that will be transmitted on Monday, see: http://bugs.digium.com/bug_view_page.php?bug_id=985 Feel free to add your own phrases. I will exclude phrases that I feel are in poor taste, or that will not (in my opinion) be used by a reasonable portion of the community. Feel free to send me a few bucks via paypal ([EMAIL PROTECTED]) to help diffuse the cost of the recordings. As usual, any payments over the amount needed to pay Allison will go to reasonably healthy food shipments to the Digium staff. Last time we had enough left over to buy a fruit basket, some pistachios, and (not so healthy) chocolate stuff, which keeps Mark etc. in the office longer and doing happier coding. :-) Please do NOT reply to the list with your words if you want me to read them; I haven't the time in the next week to filter through the list due to travel, so please update ONLY the bugs.digium.com interface with your requests. I estimate that we'll send the list on Monday, and get it back Tuesday. I'll hopefully have the phrases chopped up and in .gsm format by Wednesday and sent to Mark for inclusion in the asterisk-sounds CVS directory. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Retrieving an application command return code
Could someone please point me to the right direction on this? I'm trying to figure out how to retrieve the return code of an application command. I found this link http://lists.digium.com/pipermail/asterisk-users/2003-April/009816.html that talk about a command called "OnResultGoto", but I can't find any other mentions of that command anywhere. Has that been implemented? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel on Debian
Hi, I have Zaptel on Debian Testing with 2.4.24 kernel. I am building stuff from CVS instead of using packages for right now. I like to get my bug fixes fresh everyday or 3. Here is what I did to make it work. Had Deb testing already running. Built 2.4.18,19,20,22,24 (had zap working with all of them) Installed the kernel and reboot. Downloaded Asterisk, Libpri(not sure if this is needed for X100P but have it anyway), Zaptel fromCVS. Built them all (at first I think I had to find some dev packages such as libdb4-dev and a fewothers) with a make install If it is your first time for asterisk then also do make samples. Configured zaptel.conf like this. fxsks=1 loadzone = us defaultzone=us Loaded the wcfxo module with modprobe wcfxo Ran ztcfg (to configure the card) Set Zapata.conf like this signalling=fxs_ks rxgain=6.0 channel=1 Started Asterisk and it worked. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joe Phillips Sent: Wednesday, March 03, 2004 09:56 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] zaptel on Debian On Wed, 2004-03-03 at 00:34, Hermann Wecke wrote: > On Thu, 5 Feb 2004, Tim Sailer wrote: > > Does anyone have the zaptel modules built for Debian 2.4.24 kernel? > > Someone here is running * on debian? I have * running on Debian stable. I back-ported the zaptel and asterisk packages from testing. I'm currently evaluating * for my uses and so this is not a production system. My back-ported packages are also not complete although they are functional for my uses so far. > Recompiled kernel to latest (?) 2.4.24 > uname -a > Linux mail 2.4.24 #5 SMP Tue Mar 2 17:31:13 BRT 2004 i686 GNU/Linux stegosaurus:/etc/asterisk# uname -a Linux stegosaurus 2.4.18-1-686 #1 Mon Jan 5 19:32:08 UTC 2004 i686 unknown do you have a hardware problem? the zaptel hardware "just worked" for me. -joe -- Innovation Software Group, LLC - http://www.innovationsw.com Custom Internet and Computer Solutions Linux, UNIX, Java Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 ASN.1 Vulnerabilities: Request for "official" patch!
Sorry, didn't realise that you were the original person who asked, you should have referenced the original post saying it doesn't work and that changes are needed in ast_h323.cpp to enable people to use the new openh323. Saying asterisk has a security hole isn't really correct, it's openh323. Most asterisk people don't care about H.323 so you're at the mercy of the few that do. I'd suggest A) you ask nicely, B) make a $ bounty or C) work out the problem yourself. Have fun, Adam Jim Rosenberg wrote: See the existing discussion on this Ditto. IT DOES NOT WORK. Compiles, but no calls go through. I asked you to post your exact versions of all components, but I don't believe you did this. I have not been able to get it to work with Asterisk 0.7.2. Just because *YOU* got it to work on your particular system does not mean the problem is solved. If there is a way to get it to work reliably: 1. Please post complete details 2. Someone update asterisk.org with correct information. I believe it is correct that there is no "official" response on this from Asterisk to what many people consider a "critcal" security issue. "Read the archives" is nice, but really, the "default" Asterisk should be fixed. And the fix needs to be tested on a variety of systems, too. I tried your exact version of pwlib, and have not been able to get a *SINGLE* call to work. See the existing discussion on this Ahem. I posted pretty thorough details on what wasn't working ... Please respond so that the "discussion" can -- uh -- exist ... -T.i.A., Jim [Apologies for bandwidth-wasting inclusion below -- I'm reposting since someone thinks this discussion has been "settled" ...] On Thu, Feb 26, 2004 at 09:18:45AM +0800, [EMAIL PROTECTED] wrote: In the Makefile inside asterisk/channels/h323 directory, there's a line like this: CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include try to use "-I$(PWLIBDIR)/include" ONLY, it should work. I've compiled it with pwlib 1_6_2, which works fine leo Sigh. I am having a very rough time here. Could you please post exactly which versions of Asterisk and OpenH323 you used? When I use your advice above I get a successful build, but I haven't got a single call to actually *work* through H.323. Here are my results (all trials are Asterisk 0.7.2): OpenH323 1.13.0 / Pwlib 1.6.0: Asterisk segfaults when it gets an H.323 call. OpenH323 1.13.2 / Pwlib 1.6.3: Channel won't load, there's an unresolved symbol. OpenH323 1.13.2 / Pwlib 1.6.2: Asterisk appears to be fully stable. As far as Asterisk is concerned, everything works: calls are made, answered, bridged, all looks fine from the console. But nothing is actually making it *back* through H.323 from the Asterisk end. When I call Asterisk through H.323, Asterisk thinks things are fine, but from the calling end it thinks no one answered. When I call from the Asterisk end, I never hear anything that sounds like an answer. Now this looks *VERY* familiar. It sure is like the H.323 problems I had right at first until I caught on to using *only* G.711 A-law. Once I started making sure everyone was on ALAW, H.323 starting working fine (except for DTMF, but that's a subject for a new thread ...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 ASN.1 Vulnerabilities: Request for "official" patch!
See the existing discussion on this Ditto. IT DOES NOT WORK. Compiles, but no calls go through. I asked you to post your exact versions of all components, but I don't believe you did this. I have not been able to get it to work with Asterisk 0.7.2. Just because *YOU* got it to work on your particular system does not mean the problem is solved. If there is a way to get it to work reliably: 1. Please post complete details 2. Someone update asterisk.org with correct information. I believe it is correct that there is no "official" response on this from Asterisk to what many people consider a "critcal" security issue. "Read the archives" is nice, but really, the "default" Asterisk should be fixed. And the fix needs to be tested on a variety of systems, too. I tried your exact version of pwlib, and have not been able to get a *SINGLE* call to work. See the existing discussion on this Ahem. I posted pretty thorough details on what wasn't working ... Please respond so that the "discussion" can -- uh -- exist ... -T.i.A., Jim [Apologies for bandwidth-wasting inclusion below -- I'm reposting since someone thinks this discussion has been "settled" ...] On Thu, Feb 26, 2004 at 09:18:45AM +0800, [EMAIL PROTECTED] wrote: In the Makefile inside asterisk/channels/h323 directory, there's a line like this: CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include try to use "-I$(PWLIBDIR)/include" ONLY, it should work. I've compiled it with pwlib 1_6_2, which works fine leo Sigh. I am having a very rough time here. Could you please post exactly which versions of Asterisk and OpenH323 you used? When I use your advice above I get a successful build, but I haven't got a single call to actually *work* through H.323. Here are my results (all trials are Asterisk 0.7.2): OpenH323 1.13.0 / Pwlib 1.6.0: Asterisk segfaults when it gets an H.323 call. OpenH323 1.13.2 / Pwlib 1.6.3: Channel won't load, there's an unresolved symbol. OpenH323 1.13.2 / Pwlib 1.6.2: Asterisk appears to be fully stable. As far as Asterisk is concerned, everything works: calls are made, answered, bridged, all looks fine from the console. But nothing is actually making it *back* through H.323 from the Asterisk end. When I call Asterisk through H.323, Asterisk thinks things are fine, but from the calling end it thinks no one answered. When I call from the Asterisk end, I never hear anything that sounds like an answer. Now this looks *VERY* familiar. It sure is like the H.323 problems I had right at first until I caught on to using *only* G.711 A-law. Once I started making sure everyone was on ALAW, H.323 starting working fine (except for DTMF, but that's a subject for a new thread ...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, Mar 02, 2004 at 02:19:16PM -0500, Steve Creel wrote: > [incoming] > exten => 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX FWIW, I've done something like this and it was absolutely wonderful. We were actually running new phones and putting them in parallel with the existing system over the same phone lines (they ran 4-pair UTP to each phone jack, so we just stole the outer pair and bought some magic adapters to pull out the "second line"). You can imagine the surprise when both phones worked simultaneously. It was even more surprising when we were training people on the new phones by having them dial an "outside line" and then use the new dial codes for voicemail and such. It went over VERY well. > >How I can forward a call? It's simply an extension.conf rule? > Yes. Most people miss this. Use the Dial application (as the example shows). Dial is used for outside lines and such, but it is *'s fundamental way to make one channel dial to another. Virtually every situation where you are "forwarding" something (say Zap to SIP, SIP to IAX, TDM to TDMoe) you end up using a Dial. > >When I make the forward in this way (with extension.conf rule) asterisk > >make some work or is a simple passthrough from interfaces? > Yes, it's some switching/callsetup work, but no codec translation, which > is by far your biggest CPU consumer. It acts as a simple passthru for the CHANNELS. That is, what comes in on an individual channel goes out on another. Mapping the whole T1 would be another story (it can be done, I had to once). DACS works well for that but Asterisk can't get at the calls. I recommend the above. The only time it doesn't work well is when people want to do something with "line 5". I had a situation where certain lines couldn't dial long distance. Since the above would dynamically choose a line, it would cause unexpected problems because the old PBX's line X was no longer actually the same T1 channel on the outside. > > >I need that calls "from PRI to PRI" don't load the computer. > >I want to use all CPU to (future) SIP calls. Once the call is linked, all the load is on the Zaptel board. That is REALLY handy. I can't tell you how surprised some of my customers get when I have three machines switching 300 lines with like 5% or so load a piece. Feel free to e-mail me or jabber me (same as my e-mail address) if you have problems. I love to help set things like this up--especially in an more casual setting (you never get to have FUN with people's businesses). Jayson Vantuyl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VMware, * and SJphone ... newbie
I've read and tried a LOT of sample config's for sip.conf and extensions.conf and no matter what I do I get registration error's when trying to get SJphone registered to my * server. I have a XP VMware host with Redhat 9 / * as a guest. The SJphone is on the host XP trying to register with the guest Redhat/* server. Any suggestions? I think I'm just missing something stupid. I'm used to the idea of using VMWare so that the adult operating system is the host, and the toxic delinquent is the guest, rather than vice versa, as you have it, so this may not pertain. Check routing!!! Do you have network connectivity between the host and guest *generally*? How do you have your networking set up? When Linux is the host, guests can be set up in a variety of configurations -- NAT, bridge, etc. With Linux as the host, routed can be somewhat cranky at routing from your host LAN to NATted VMs. Can you ping your host from the guest? This may not be an Asterisk issue at all. It might be a networking issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 ASN.1 Vulnerabilities: Request for "official" patch!
See the existing discussion on this - basically download compile the new pwlib & openh323 and recompile channels/h323 - you'll need to remove -Isomething/unix from the Makefile Jim Rosenberg wrote: To recap: 1. Security vulnerabilities have been found in the ASN.1 parsing of *many* H.323 implementations. Some security experts consider them quite serious, others don't. 2. OpenH323 *was* vulnerable when the announcement was made. (About a month and a half ago, or so.) 3. The OpenH323 folks patched their code quite quickly. I belive that to obtain their fix you need to check code out of CVS. 4. If you visit asterisk.org, follow "the usual" download instructions, and build in H.323 support, your resulting Asterisk *WILL* be vulnerable. 5. Integrating a "fixed" version of OpenH323 with Asterisk is not straightforward. (I at least have not been able to get this to work.) 6. There is (in my opinion) *widespread misunderstanding* on this issue. E.g., I had Digium support try to convince me that Asterisk was not vulnerable. I would like to make a public appeal to whoever is in position to do this to issue an "official" patch -- and to update the asterisk.org website so newbies get a fixed version when they download and build in H.323 support. Please please please ... -T.i.A., Jim ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 ASN.1 Vulnerabilities: Request for "official" patch!
To recap: 1. Security vulnerabilities have been found in the ASN.1 parsing of *many* H.323 implementations. Some security experts consider them quite serious, others don't. 2. OpenH323 *was* vulnerable when the announcement was made. (About a month and a half ago, or so.) 3. The OpenH323 folks patched their code quite quickly. I belive that to obtain their fix you need to check code out of CVS. 4. If you visit asterisk.org, follow "the usual" download instructions, and build in H.323 support, your resulting Asterisk *WILL* be vulnerable. 5. Integrating a "fixed" version of OpenH323 with Asterisk is not straightforward. (I at least have not been able to get this to work.) 6. There is (in my opinion) *widespread misunderstanding* on this issue. E.g., I had Digium support try to convince me that Asterisk was not vulnerable. I would like to make a public appeal to whoever is in position to do this to issue an "official" patch -- and to update the asterisk.org website so newbies get a fixed version when they download and build in H.323 support. Please please please ... -T.i.A., Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] on hold music from a mp3 stream or sound card input?
Hi Folks, Rather than have my hold music play from a sound file I'd like to have a live feed from a sound card input or MP3 stream. Is this doable and if so how? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
>> Perhaps I should have posted my question differently to the list: After installing the CVS version of Asterisk, I type, "modprobe xct1xxp.">> The machine accepts the command but the LED on the T100P does not flash.>> How do I know that the T100P module has loaded correctly? Do you see any errors showing up in the output from "dmesg"? __James, That output looks good. The last three lines read, "Zapata Telephony Interface Registered on major 196Framer: DS21552, Revision: 3 (T1)Found a Wildcard: Digium Wildcard T100P T1/PRI" chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has "t" or "T" in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
Use default configuration as below should work if you have PRI line. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=default switchtype=national signalling=pri_cpe channel=>1-23 group = 1 If it's not green, make sure you put in right framing, coding, signalling, switchtype. Call your telco and ask them if you do not know for sure. It could be E&M Wink (channel bank line) which uses different settings. Hope this help. Tools: /sbin/ztcfg or /sbin/zttool -Tri. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 03, 2004 3:28 PM Subject: Re: [Asterisk-Users] wct1xxp module and the T100P Andrew McRory <[EMAIL PROTECTED]> Sent by: [EMAIL PROTECTED] 03/03/2004 04:11 PM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: Re: [Asterisk-Users] wct1xxp module and the T100POn Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote:> I'm having trouble turning up a PRI to a T100P. I've read on the Digium > FAQ's that once the wct1xxp module is loaded correctly, the LED on the > T100P will flash red. I believe I've loaded the module correctly because > both wct1xxp and zaptel are listed when I do the "lsmod" command. The LED > on the card does not flash on and off. Does anyone have any > recommendations on what I could be doing wrong?Switch type, line code, framing all matter. How about posting your config?-- Andrew McRory - President/CTOLinux Systems Engineers, Inc.PO BOX 3791Tallahassee, FL 32315(850)224-5737(850)294-7567___ Sure: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=cyperpri switchtype=national pridialplan=unknown signalling=pri_cpe channel=>1-23 Here's a question, though: does the wct1xxp module read from either zaptel.conf or zapata.conf when loaded? Thanks! chris
Re: [Asterisk-Users] ENUM when your country's ITU representative is uncooperative
Stephen Davies wrote: Considering that they probably won't delegate, how about Asterisk supporting a second parallel ENUM tree under a domain that we can control ourselves? http://e164.freenetworks.org See my previous posts about this... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
> Steven, > > Perhaps I should have posted my question differently to the list: > > After installing the CVS version of Asterisk, I type, "modprobe xct1xxp." > The machine accepts the command but the LED on the T100P does not flash. > How do I know that the T100P module has loaded correctly? Do you see any errors showing up in the output from "dmesg"? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
> > On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: > > I'm having trouble turning up a PRI to a T100P. I've read on the > > Digium FAQ's that once the wct1xxp module is loaded correctly, the > LED > > on the T100P will flash red. I believe I've loaded the module > > correctly because both wct1xxp and zaptel are listed when I do the > > "lsmod" command. The LED on the card does not flash on and off. > Does > > anyone have any recommendations on what I could be doing wrong? > > Thanks! > > chris > > > > Could be wrong or bad cable, could be incorrect configuration, could > be > no service on the line yet. > > Must provide data to get information. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > > __ > > I know the PRI is good to the RJ-45 right before it goes into the > T100P; I've made calls on it with our T-Berd. So I'm sure the cable > is ok and there is service on the line. > > I haven't run Asterisk yet; shouldn't the card look "alive" once its > driver module is loaded? > Thanks! > Chris Nope, What would answer the request placed on the line. BTW, please see about putting some sane configuration options in your mailer. I know Lotus is notoriously crappy when viewed by other readers, but that was really bad. -- Steven Critchfield <[EMAIL PROTECTED]> __ Steven, Perhaps I should have posted my question differently to the list: After installing the CVS version of Asterisk, I type, "modprobe xct1xxp." The machine accepts the command but the LED on the T100P does not flash. How do I know that the T100P module has loaded correctly? (What sort of mailer config options do you recommend?) Thanks! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
Man your email client is borked! zaptel looks good. try removing pridialplan=unknown and add group=1. -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
So, sorry I have general question , h.323 dont work on FreeBSD + asterisk ???,,, I need converter h.323 <> sip and codec converter for h.323. I use FreeBSD 5.2. > Thanks all, Serge. - Original Message - > From: "NetOne Administrator" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 01, 2004 11:09 PM > Subject: Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP ! > > > > FreeBSD asterisk port is *NUTS* > > Don't use it! > > > > Asterisk compiles just fine on BSD, if you are using 4.x-RELEASE, and > > not using chan_h323, chan_oss, zaptel & libpri. > > > > Darren Wiebe wrote: > > > > > Sorry to just come on line now. Have you tried the FreeBSD port? > > > net/asterisk is the place to look. It always dumps core on me but you > > > may have better luck. > > > > > > Darren Wiebe > > > [EMAIL PROTECTED] > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
Andrew McRory <[EMAIL PROTECTED]> Sent by: [EMAIL PROTECTED] 03/03/2004 04:11 PM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: Re: [Asterisk-Users] wct1xxp module and the T100P On Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote: > I'm having trouble turning up a PRI to a T100P. I've read on the Digium > FAQ's that once the wct1xxp module is loaded correctly, the LED on the > T100P will flash red. I believe I've loaded the module correctly because > both wct1xxp and zaptel are listed when I do the "lsmod" command. The LED > on the card does not flash on and off. Does anyone have any > recommendations on what I could be doing wrong? Switch type, line code, framing all matter. How about posting your config? -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ Sure: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=cyperpri switchtype=national pridialplan=unknown signalling=pri_cpe channel=>1-23 Here's a question, though: does the wct1xxp module read from either zaptel.conf or zapata.conf when loaded? Thanks! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 2004-03-03 at 17:16, [EMAIL PROTECTED] wrote: > > > > Steven Critchfield > <[EMAIL PROTECTED]> > Sent by: > [EMAIL PROTECTED] > > 03/03/2004 04:06 PM > Please respond to > asterisk-users > > To: > [EMAIL PROTECTED] > cc: > Subject: > Re: [Asterisk-Users] > wct1xxp module and the > T100P > > > On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: > > I'm having trouble turning up a PRI to a T100P. I've read on the > > Digium FAQ's that once the wct1xxp module is loaded correctly, the > LED > > on the T100P will flash red. I believe I've loaded the module > > correctly because both wct1xxp and zaptel are listed when I do the > > "lsmod" command. The LED on the card does not flash on and off. > Does > > anyone have any recommendations on what I could be doing wrong? > > Could be wrong or bad cable, could be incorrect configuration, could > be > no service on the line yet. > > Must provide data to get information. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > > __ > > I know the PRI is good to the RJ-45 right before it goes into the > T100P; I've made calls on it with our T-Berd. So I'm sure the cable > is ok and there is service on the line. > > I haven't run Asterisk yet; shouldn't the card look "alive" once its > driver module is loaded? Nope, What would answer the request placed on the line. BTW, please see about putting some sane configuration options in your mailer. I know Lotus is notoriously crappy when viewed by other readers, but that was really bad. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
Steven Critchfield <[EMAIL PROTECTED]> Sent by: [EMAIL PROTECTED] 03/03/2004 04:06 PM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: Re: [Asterisk-Users] wct1xxp module and the T100P On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: > I'm having trouble turning up a PRI to a T100P. I've read on the > Digium FAQ's that once the wct1xxp module is loaded correctly, the LED > on the T100P will flash red. I believe I've loaded the module > correctly because both wct1xxp and zaptel are listed when I do the > "lsmod" command. The LED on the card does not flash on and off. Does > anyone have any recommendations on what I could be doing wrong? Could be wrong or bad cable, could be incorrect configuration, could be no service on the line yet. Must provide data to get information. -- Steven Critchfield <[EMAIL PROTECTED]> __ I know the PRI is good to the RJ-45 right before it goes into the T100P; I've made calls on it with our T-Berd. So I'm sure the cable is ok and there is service on the line. I haven't run Asterisk yet; shouldn't the card look "alive" once its driver module is loaded? Thanks! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote: > I'm having trouble turning up a PRI to a T100P. I've read on the Digium > FAQ's that once the wct1xxp module is loaded correctly, the LED on the > T100P will flash red. I believe I've loaded the module correctly because > both wct1xxp and zaptel are listed when I do the "lsmod" command. The LED > on the card does not flash on and off. Does anyone have any > recommendations on what I could be doing wrong? Switch type, line code, framing all matter. How about posting your config? -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: > I'm having trouble turning up a PRI to a T100P. I've read on the > Digium FAQ's that once the wct1xxp module is loaded correctly, the LED > on the T100P will flash red. I believe I've loaded the module > correctly because both wct1xxp and zaptel are listed when I do the > "lsmod" command. The LED on the card does not flash on and off. Does > anyone have any recommendations on what I could be doing wrong? Could be wrong or bad cable, could be incorrect configuration, could be no service on the line yet. Must provide data to get information. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wct1xxp module and the T100P
I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the "lsmod" command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong? Thanks in advance! chris
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? -- Zen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] KPN BRI
This is a bit of a longshot, but do you know if your line is configured point-to-point to point-to-multipoint. We found with netjet cards in australia they only worked if the line was configured point-to-multipoint and funny things happened if the line wasn't. We could dial out of the line with a ISDN phone alright just not with the PBX. Another thing is check your msn details. Regards, Matthew Enger [EMAIL PROTECTED] On Thu, 2004-03-04 at 02:12, Mark wrote: > > The software configuration depends (of course) on your hardware > > configuration. In this situation, what kind of hardware do you plan to > > use to connect your asstricks box to KPN? You are best of going > > with a capi capable isdn card, and using chan_capi (see > > http://www.junghanns.net/asterisk/page1.html), your > > alternative would be to use chan_modem together with isdn4linux. > > I've done both, so my is based on personal experience. > > I have 2 Eicon Diva cards which I am using chan_capi. > > I have chan_capi installed and configured and it detects the ports ok. > > I have the lines plugged in but when I dial the number associated with the > line does not get picked up and I get a non-existant number tone. When I plug > in a standard isdn telephone into the line it all works ok. > > I think I am using the wrong kind of signalling. I have found out that kpn use > e164 as the signalling but I cannot find anywhere to configure this. > > Thanks for any advice you can give. > > Regards > > Mark > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger <[EMAIL PROTECTED]> Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco VIP30
Hi Just got a brand new Box Cisco VIP30 off ebay, the standard phone functions work fine, just a couple of questions, 1) how do I program the other buttons not on the standard keypad part.. 2) When I hang up the display doesn't clear and keeps the numbers just dialed on screen, can this be cleared down. thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold with Pingtel?
On Wed, 3 Mar 2004, Nate Carlson wrote: > I'd like to get Music on Hold working (so when I hit the 'Hold' button > on my pingtel phone the caller gets music until I pick the phone back > up); I can't seem to find any sample config files on how to do this. > I've tested the music on hold subsystem using: > > exten => 3000,1,Answer > exten => 3000,2,MusicOnhold(loud) > > ..and that works fine (so I know I've got mpg123 set up properly and > stuff), but it doesn't work when I call my pingtel and then hit hold. > Any ideas? Thanks! Never mind - figured it out; just had to uncomment the 'default' section in musiconhold.conf, and then it worked automagically. D'oh! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Setup and configuration help
Hi, I have a client that wants to set up an Asterisk-based VOIP solution. While we can easily handle most of their IT needs, we've never really used Asterisk. We're looking for an experienced Asterisk tech to give us some help getting the box up and running, configured appropriately. Once up and running, I imagine that this person would give us a little training and provide third-level support as required. If interested, please email me off list - make sure to put "Asterisk" in the subject line so I don't mistake it for spam. Regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium CVS Server: Connection refused?
> I've been having trouble getting updates recently, but it does eventually > go through. > Thanks. It finally went through for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium CVS Server: Connection refused?
On Wed, 2004-03-03 at 16:06, Steven Sokol wrote: > I seem to be having problems doing an update from Digium's CVS. Has anybody > heard anything? I got no response from anybody on the IRC channel. Check your DNS. Verify you are using DNS. Earlier this week(I think) there was 2 servers in the return for a DNS lookup for cvs.digium.com. Now there is just the one entry that was working from when that was a problem. If you are using an old IP, or have old DNS records cached, you will experience trouble. BTW, I have no trouble connecting. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium CVS Server: Connection refused?
At 02:06 PM 3/3/2004, you wrote: I seem to be having problems doing an update from Digium's CVS. Has anybody heard anything? I got no response from anybody on the IRC channel. I've been having trouble getting updates recently, but it does eventually go through. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P UK PRI Configuration
> We have 1 ISDN PRI (E1) provided by BT (UK) for incoming and outgoing calls. > > Incoming calls work fine and the are no alarms on back of card or in > /proc/zaptel/1, but with outgoing calls, > all numbers are rejected with the BT error "The number you have dialed has > not been recognized, please check and try again" Assuming that you are actually dialling the numbers correctly with your extensions.conf / Dial commands, then you might need to experiment with: ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN pridialplan=national in /etc/asterisk/zapata.conf - this is most likely what is going wrong. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium CVS Server: Connection refused?
I seem to be having problems doing an update from Digium's CVS. Has anybody heard anything? I got no response from anybody on the IRC channel. Cheers, Steven Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA 186 Firmware
v.3.0 works fine too James Coberly wrote: v2.16.2 ata18x Works Fine for me. - Original Message - From: "Erick Weber V." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 1:18 PM Subject: [Asterisk-Users] Best ATA 186 Firmware Hi: Someone know wich is the best firmware for the ATA 186 with * Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P UK PRI Configuration
check with bt what kind of dialplan they use and set it with pridialplan= in zapata.conf On Wednesday 03 of March 2004 22:08, Michael East wrote: > fine and the are no alarms on back of card or in > /proc/zaptel/1, but with outgoing calls, > all numbers are rejected with the BT error "The number you have dialed has > not been recognized, please check and try again" > > Outgoing calls work fine though the SDX when the PRI is unplugged from the > asterisk server and plugged back into the SDX, so it rules out a problem > with the PRI. > > I think I have misconfigured somthing somewhere, anyone got any ideas ? > > Below are the zapata/zaptel configs we are using > > zaptel.conf > span=1-8,1,0,ccs,hdb3,crc4 > indclear=1-15 > dchan=16 > indclear=17-31 > defaultzone=uk > > zapata.conf > [channels] > callwaiting=no > threewaycalling=no > transfer=yes > cancallforward=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=yes > rxgain=0.0 > txgain=0.0 > switchtype=euroisdn > callerid = asreceived > group=1 > context=default > signalling=pri_cpe > channel = 1-8 > > Regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ENUM when your country's ITU representative is uncooperative
On Wed, Mar 03, 2004 at 10:40:09PM +0200, Stephen Davies wrote: > So - what to do? If I approach the administrators for e164.arpa > ([EMAIL PROTECTED], apparently) will they delegate 7.2.e164.arpa > to me? > I guess that they won't. (It would be fun if they > would, for some definition of fun (I once administered .mu > and the Mauritius telco thought THEY should administer it)). Very unlikely, RIPE will only delegate to a "governmental approved" organisation. > Considering that they probably won't delegate, how about Asterisk > supporting a second parallel ENUM tree under a domain that we can > control ourselves? Have alook at freenum.org Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P UK PRI Configuration
Hi, We have just upgraded from a Sdx Index V200 PBX to Asterisk and are having a few problems. We have 1 ISDN PRI (E1) provided by BT (UK) for incoming and outgoing calls. Incoming calls work fine and the are no alarms on back of card or in /proc/zaptel/1, but with outgoing calls, all numbers are rejected with the BT error "The number you have dialed has not been recognized, please check and try again" Outgoing calls work fine though the SDX when the PRI is unplugged from the asterisk server and plugged back into the SDX, so it rules out a problem with the PRI. I think I have misconfigured somthing somewhere, anyone got any ideas ? Below are the zapata/zaptel configs we are using zaptel.conf span=1-8,1,0,ccs,hdb3,crc4 indclear=1-15 dchan=16 indclear=17-31 defaultzone=uk zapata.conf [channels] callwaiting=no threewaycalling=no transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 switchtype=euroisdn callerid = asreceived group=1 context=default signalling=pri_cpe channel = 1-8 Regards, Michael East ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ENUM when your country's ITU representative is uncooperative
On Wed, 2004-03-03 at 14:40, Stephen Davies wrote: > Hi, > > I thought it would be neat to put my SIP/IAX reachable systems into > the ENUM system. > > But reading about it I see that its rather centrally controlled within > the ITU. > > My country code (+27) is not delegated. My country has a monopoly > telco whose only interest in VOIP is to keep it all to themselves and > not permit any other usage. > > So - what to do? If I approach the administrators for e164.arpa > ([EMAIL PROTECTED], apparently) will they delegate 7.2.e164.arpa > to me? > > I guess that they won't. (It would be fun if they > would, for some definition of fun (I once administered .mu > and the Mauritius telco thought THEY should administer it)). > > Considering that they probably won't delegate, how about Asterisk > supporting a second parallel ENUM tree under a domain that we can > control ourselves? Hmm, isn't that pretty much what iaxtel and FWD do currently except without DNS in the way? Basically you are looking for a way for you to get a few addresses/phone numbers pointed to your VoIP system, and FWD will do that, and so will Iaxtel. Granted you will then be requiring your contacts to also use them, but seems that enum is the same way too. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Deleting old voicemail messages
On Wednesday 03 March 2004 14:34, Jim Sneeringer wrote: > It would be nice to have an option to delete voicemail messages > that reached a certain age, or to delete those that are delivered > by e-mail. I gather from searching previous posts that this has not > yet been done, and the solution is to set up a chron job. contrib/scripts/messages-expire.pl contrib/scripts/README.messages-expire -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and Voicemail Memory increasing
On Wed, 2004-03-03 at 14:36, Robert Boardman wrote: > Steven Critchfield wrote: > > >On Wed, 2004-03-03 at 06:33, Robert Boardman wrote: > > > > > >>Hi, > >> > >>With the current CVS as of last night 20:00GMT > >> > >>I was testing a asterisk with the e100p card using a PRI analyzer to excerise > >>the 30 channels over and over, just going directly to voice-mail. > >> > >>Basically, I don't know what is going on but every time a PRI channel is picked > >>up about 8K memory is used, but it is never released, I ran the test for about > >>four hours last night, and th server started with 100Mb usage, and after 4 hours > >>the memory usage had climbed to 270Mb. > >> > >>After some research and a few changes this is my current observation > >> > >>1st Observation: > >>After the memory leak was discovered I turned astmm on and recompiled *. The > >>'show memory allocations' and 'show memory summary' commands did not > >>indicate that any of the * internals were growing during the test run. > >> > >>2nd Observation: > >>When I first boot the system (mod probes for zaptel and wct1xxp and * are run up > >>from rc.local and there are no ISDN calls running) the memory is already leaking > >>by 8Kb every few seconds. If I stop * and then restart it the "memory leak" > >>stops until I start making ISDN calls... > >> > >>Does anyone have any advice to release the memory or stop the leak? > >> > >> > > > >Where are you getting the values for the memory usage? Are you just > >using the free memory on the system? > > > > > yes, Im just really looking at TOP memory usage when asterisk is being > hammered by the pri analyzer > > Is this just liunx using memory for buffers and cache?? Probably. 8k every few seconds sounds exactly like cache and buffers. Remember that until physical RAM is exhausted, buffers and cache will grow to fill all RAM. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ENUM when your country's ITU representative is uncooperative
Hi, I thought it would be neat to put my SIP/IAX reachable systems into the ENUM system. But reading about it I see that its rather centrally controlled within the ITU. My country code (+27) is not delegated. My country has a monopoly telco whose only interest in VOIP is to keep it all to themselves and not permit any other usage. So - what to do? If I approach the administrators for e164.arpa ([EMAIL PROTECTED], apparently) will they delegate 7.2.e164.arpa to me? I guess that they won't. (It would be fun if they would, for some definition of fun (I once administered .mu and the Mauritius telco thought THEY should administer it)). Considering that they probably won't delegate, how about Asterisk supporting a second parallel ENUM tree under a domain that we can control ourselves? Thanks for any comments, Steve Davies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and Voicemail Memory increasing
Steven Critchfield wrote: On Wed, 2004-03-03 at 06:33, Robert Boardman wrote: Hi, With the current CVS as of last night 20:00GMT I was testing a asterisk with the e100p card using a PRI analyzer to excerise the 30 channels over and over, just going directly to voice-mail. Basically, I don't know what is going on but every time a PRI channel is picked up about 8K memory is used, but it is never released, I ran the test for about four hours last night, and th server started with 100Mb usage, and after 4 hours the memory usage had climbed to 270Mb. After some research and a few changes this is my current observation 1st Observation: After the memory leak was discovered I turned astmm on and recompiled *. The 'show memory allocations' and 'show memory summary' commands did not indicate that any of the * internals were growing during the test run. 2nd Observation: When I first boot the system (mod probes for zaptel and wct1xxp and * are run up from rc.local and there are no ISDN calls running) the memory is already leaking by 8Kb every few seconds. If I stop * and then restart it the "memory leak" stops until I start making ISDN calls... Does anyone have any advice to release the memory or stop the leak? Where are you getting the values for the memory usage? Are you just using the free memory on the system? yes, Im just really looking at TOP memory usage when asterisk is being hammered by the pri analyzer Is this just liunx using memory for buffers and cache?? Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Deleting old voicemail messages
It would be nice to have an option to delete voicemail messages that reached a certain age, or to delete those that are delivered by e-mail. I gather from searching previous posts that this has not yet been done, and the solution is to set up a chron job. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segfault when parking from extension dialed inside AGI.
Hello again
After a few minutes of thinking (usefull sometimes :) I solved the
problem of using the AGI to make the dialing decision while avoid doing
the dial from inside the agi application without changing context (to
keep access to other extensions using transfer).
Very simple, using SET PRIORITY together with SET VARIABLE after I get
the needed information,
and then exiting from agi before doing the Dial.
Something like this:
[macro-generic_dial]
; ${ARG1} - Extension
;
exten => s,1,AGI(gotodial.agi,${ARG1})
exten => s,2,Hangup
exten => s,3,Dial(${DIALCHANNEL},40,tr) ; Ring the selected channel 40
seconds
exten => s,4,GotoIf($[${VMBOX} = 0]?4:3)
exten => s,5,VoiceMail([EMAIL PROTECTED]) ; If unavailable, send to
voicemail
exten => s,6,Hangup ; Hangup the call
exten => s,104,GotoIf($[${VMBOX} = 0]?104:103)
exten => s,105,VoiceMail([EMAIL PROTECTED]) ; If busy, send to voicemail
exten => s,106,Hangup ; Hangup the call
[default]
;; Generic extention dialing
exten => _1XX,1,Macro(generic_dial,${EXTEN})
And using something like this inside the gotodial.agi script:
.
$agi->database_put("LastDial", $callinExten,
$exten); # To implement call return
$agi->verbose("Dialing: $dialstring",1);
$agi->set_variable('DIALCHANNEL',"$dialstring");
$agi->set_variable('VMBOX',"$vm");
$agi->set_priority('3');
exit 0;
..
Anyway It would be nice If asterisk didn't die so easily (h323 transfer,
agi dial and parking, openswitch channel's driver, speaking too loudly
next to the server, etc)
Best regards
Luis
Luis Vazquez wrote:
Hello all,
Asterisk is segfault dying when I try to park a call from an extension
dialed from an AGI script.
The situation is as follows:
I call from a sip phone (really It doesn't matter if It's SIP or not)
to extension 181 (corresponding to a mgcp DG-104S phone).
.
exten => 181,1,AGI(dummydial.agi,MGCP/aaln/[EMAIL PROTECTED])
.
dummydial.agi is only a simplified test script I made to isolate the
problem, It only makes a dial to the given channel:
#
#!/usr/bin/perl
# dummydial.agi: Marcar internos con AGI
use strict;
use Asterisk::AGI;
local $::INPUT_RECORD_SEPARATOR="\n";
local $::OUTPUT_AUTOFLUSH=1;
my $agi = new Asterisk::AGI;
my %input = $agi->ReadParse();
my ($dialstring) = shift @ARGV;
$agi->verbose("Dialing: $dialstring",1);
my $ret = $agi->exec('Dial',"$dialstring|40|t");
# $ret = $agi->exec('Macro',"generic_dial|$dialstring|$vm"); # This
is the real thing, crashes to
# exit 0;
##
then I peak the phone and the call is established perfectly. Then I
want to park the call with the following configuration at parking.conf:
###
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park
calls on
context => parkedcalls ; Which context parked calls
are in
parkingtime => 120 ; Number of seconds a call can
be parked
so from the called extension (181) I press (#) I do hear "transfer"
and then I dial 700.
Then the voice tell mi the call is parked at extension 701, but as
soon as I hangup the called phone
and try to peak the parked call (and sometimes even before) Asterisk
dies with segmentation fault.
As I said before It doesn't depend if I made a call from SIP to SIP or
MGCP to SIP or MGCP to MGCP extension.
The final result (the server crashing with segfault) is always the same.
However If I do exactly the same but with the extension dialing
directly from extensions.conf with:
exten => 183,1,Dial(MGCP/aaln/[EMAIL PROTECTED],40,tr)
the parking and recovering of the calls works correctly without any
problem.
This is the output on the console:
=
*CLI> -- Executing AGI("SIP/ipcontact.com.uy-0817d0c0",
"dummydial.agi|MGCP/aaln/[EMAIL PROTECTED]") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dummydial.agi
dummydial.agi|MGCP/aaln/[EMAIL PROTECTED]: Dialing: MGCP/aaln/[EMAIL PROTECTED]
-- AGI Script Executing Application: (Dial) Options:
(MGCP/aaln/[EMAIL PROTECTED]|40|t)
-- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: -1, dnd: 0, so: 0, sno: 0
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP/aaln/[EMAIL PROTECTED] answered SIP/ipcontact.com.uy-0817d0c0
-- Attempting native bridge of SIP/ipcontact.com.uy-0817d0c0 and
MGCP>/aaln/[EMAIL PROTECTED]
Ma
[Asterisk-Users] Professional Text to Speech
Title: Professional Text to Speech Hi everyone, I was wondering about peoples experiences with professional text to speech packages, and their integration with Asterisk? Festival at least in the default install sounds... less than perfect. Any suggestions? Matt
[Asterisk-Users] Music on Hold with Pingtel?
I'd like to get Music on Hold working (so when I hit the 'Hold' button on my pingtel phone the caller gets music until I pick the phone back up); I can't seem to find any sample config files on how to do this. I've tested the music on hold subsystem using: exten => 3000,1,Answer exten => 3000,2,MusicOnhold(loud) ..and that works fine (so I know I've got mpg123 set up properly and stuff), but it doesn't work when I call my pingtel and then hit hold. Any ideas? Thanks! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ringing Delay
Yup, that was it. Set usecallerid=no and it rings right out.
Thanks
On Wed, 2004-03-03 at 18:32, Eric Wieling wrote:
> Chances are it's waiting to get the caller ID info (sent between the
> first and the second ring)
>
> On Wed, 2004-03-03 at 12:01, WipeOut wrote:
> > Brian Mulligan wrote:
> >
> > >Sorry if this is a daft question but when a PSTN call comes in on my
> > >X100P the console shows the following;
> > >
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >-- Executing Dial("Zap/1-1", "Zap/2") in new stack
> > >-- Called 2
> > >-- Zap/2-1 is ringing
> > >-- Zap/2-1 is ringing
> > >WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
> > >Didn't finish Caller-ID spill. Cancelling.-- Zap/2-1 is ringing
> > >-- Zap/2-1 answered Zap/1-1
> > >-- Attempting native bridge of Zap/1-1 and Zap/2-1
> > >-- Hungup 'Zap/2-1'
> > > == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
> > >-- Hungup 'Zap/1-1'
> > >
> > >
> > >AS indicated, the call is switched to the context [incoming] which is
> > >configured as follows;
> > >
> > >[incoming]
> > >exten=> s,1,Dial,Zap/2
> > >
> > >The extension rings but not until the incoming line has rung three
> > >times. If I hang-up the external line before the extension is answered
> > >then the extension continues to ring three more times. Clearly, if I
> > >pick up the extension during this time then nobody is there!
> > >
> > >I would like the extension to ring immediately when the call comes in.
> > >Despite my best efforts I cannot find a configuration element which
> > >addresses it. Could it be my hardware just being too slow?
> > >
> > >Thanks
> > >Brian
> > >
> > >
> > >
> > Brian,
> >
> > AFAIK the delay is caused by the way the X100P detects ringing.. If you
> > are using digital lines (ISDN) then there is a signal to tell the device
> > connected to the line that it is ringing but with an analog line this is
> > not the case..
> >
> > So the X100P basically looks for the "swings" on the line that indicate
> > that it is ringing, it goes through about 3 of them before it answers to
> > avoid "phantom" calls which used to happen a lot and were very
> > irritating especially in the middle of the night.. So what you are
> > experiencing is normal..
> >
> > Later..
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA 186 Firmware
v2.16.2 ata18x Works Fine for me. - Original Message - From: "Erick Weber V." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 1:18 PM Subject: [Asterisk-Users] Best ATA 186 Firmware > Hi: > > Someone know wich is the best firmware for the ATA 186 with * > > Thanks > > Erick > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segfault when parking from extension dialed inside AGI.
Hello all,
Asterisk is segfault dying when I try to park a call from an extension
dialed from an AGI script.
The situation is as follows:
I call from a sip phone (really It doesn't matter if It's SIP or not) to
extension 181 (corresponding to a mgcp DG-104S phone).
.
exten => 181,1,AGI(dummydial.agi,MGCP/aaln/[EMAIL PROTECTED])
.
dummydial.agi is only a simplified test script I made to isolate the
problem, It only makes a dial to the given channel:
#
#!/usr/bin/perl
# dummydial.agi: Marcar internos con AGI
use strict;
use Asterisk::AGI;
local $::INPUT_RECORD_SEPARATOR="\n";
local $::OUTPUT_AUTOFLUSH=1;
my $agi = new Asterisk::AGI;
my %input = $agi->ReadParse();
my ($dialstring) = shift @ARGV;
$agi->verbose("Dialing: $dialstring",1);
my $ret = $agi->exec('Dial',"$dialstring|40|t");
# $ret = $agi->exec('Macro',"generic_dial|$dialstring|$vm"); # This is
the real thing, crashes to
# exit 0;
##
then I peak the phone and the call is established perfectly.
Then I want to park the call with the following configuration at
parking.conf:
###
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 120 ; Number of seconds a call can
be parked
so from the called extension (181) I press (#) I do hear "transfer" and
then I dial 700.
Then the voice tell mi the call is parked at extension 701, but as soon
as I hangup the called phone
and try to peak the parked call (and sometimes even before) Asterisk
dies with segmentation fault.
As I said before It doesn't depend if I made a call from SIP to SIP or
MGCP to SIP or MGCP to MGCP extension.
The final result (the server crashing with segfault) is always the same.
However If I do exactly the same but with the extension dialing directly
from extensions.conf with:
exten => 183,1,Dial(MGCP/aaln/[EMAIL PROTECTED],40,tr)
the parking and recovering of the calls works correctly without any problem.
This is the output on the console:
=
*CLI> -- Executing AGI("SIP/ipcontact.com.uy-0817d0c0",
"dummydial.agi|MGCP/aaln/[EMAIL PROTECTED]") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dummydial.agi
dummydial.agi|MGCP/aaln/[EMAIL PROTECTED]: Dialing: MGCP/aaln/[EMAIL PROTECTED]
-- AGI Script Executing Application: (Dial) Options:
(MGCP/aaln/[EMAIL PROTECTED]|40|t)
-- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: -1, dnd: 0, so: 0, sno: 0
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP/aaln/[EMAIL PROTECTED] answered SIP/ipcontact.com.uy-0817d0c0
-- Attempting native bridge of SIP/ipcontact.com.uy-0817d0c0 and
MGCP>/aaln/[EMAIL PROTECTED]
Mar 3 16:07:42 NOTICE[458781]: rtp.c:264 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '#'
-- Started music on hold, class 'default', on
SIP/ipcontact.com.uy-0817d0c0
-- Playing 'pbx-transfer' (language 'en')
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Stopped music on hold on SIP/ipcontact.com.uy-0817d0c0
-- Started music on hold, class 'default', on
SIP/ipcontact.com.uy-0817d0c0
== Parked SIP/ipcontact.com.uy-0817d0c0 on 701
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
-- AGI Script dummydial.agi completed, returning 0
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
-- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed
-- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-)
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '1'
-- Executing ParkedCall("MGCP/aaln/[EMAIL PROTECTED]", "701") in new stack
-- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED]
-- Stopped music on hold on SIP/ipcontact.com.uy-0817d0c0
-- Channel MGCP/aaln/[EMAIL PROTECTED] connected to parked call 701
-- Attempting native bridge of SIP/ipcontact.com.uy-0817d0c0 and
MGCP/aaln/[EMAIL PROTECTED]
Mar 3 16:07:53 WARNING[475166]: channel.c:846 ast_waitfor_nandfds:
Thread 475166 Blocking 'SIP
Re: [Asterisk-Users] VMware, * and SJphone ... newbie
On Wed, 2004-03-03 at 12:34, Jennings, Mike wrote: > I've read and tried a LOT of sample config's for sip.conf and > extensions.conf and no matter what I do I get registration error's > when trying to get SJphone registered to my * server. I have a XP > VMware host with Redhat 9 / * as a guest. The SJphone is on the host > XP trying to register with the guest Redhat/* server. > > Any suggestions? I think I'm just missing something stupid. You could have used the $300 to buy a cheap computer to run asterisk on bare hardware. Thats the first problem. Second problem is lacking details, as in error messages, when you ask a question. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sorry about the post, meant to be off-list not on.
Hit reply but did not change address!! None the less, I understand your point and respect it. All I ask is for the same respect. Alex Message: 6 From: Tilghman Lesher <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs Date: Wed, 3 Mar 2004 12:03:50 -0600 Reply-To: [EMAIL PROTECTED] On Wednesday 03 March 2004 11:47, Alex Lopez wrote: > I will respond off list, to conserve bandwidth. Which you didn't do. > I feel that this is VERY ON-TOPIC. If this continues, we will be > faced with customers shying away from Linux due to the whole FUD > factor (fear, Uncertainty, and Doubt). Then post it to a generic Linux list. This list is about using Asterisk specfically, not Linux generally. There are more than enough lists out there for which such a post is on-topic. Please take your non-Asterisk posts there. I don't think there's a single user on this list who cares about the SCO issue who hadn't already heard this news elsewhere. In any case, this list does not need to be the central ground for general Linux news. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and Voicemail Memory increasing
On Wed, 2004-03-03 at 06:33, Robert Boardman wrote: > Hi, > > With the current CVS as of last night 20:00GMT > > I was testing a asterisk with the e100p card using a PRI analyzer to excerise > the 30 channels over and over, just going directly to voice-mail. > > Basically, I don't know what is going on but every time a PRI channel is picked > up about 8K memory is used, but it is never released, I ran the test for about > four hours last night, and th server started with 100Mb usage, and after 4 hours > the memory usage had climbed to 270Mb. > > After some research and a few changes this is my current observation > > 1st Observation: > After the memory leak was discovered I turned astmm on and recompiled *. The > 'show memory allocations' and 'show memory summary' commands did not > indicate that any of the * internals were growing during the test run. > > 2nd Observation: > When I first boot the system (mod probes for zaptel and wct1xxp and * are run up > from rc.local and there are no ISDN calls running) the memory is already leaking > by 8Kb every few seconds. If I stop * and then restart it the "memory leak" > stops until I start making ISDN calls... > > Does anyone have any advice to release the memory or stop the leak? Where are you getting the values for the memory usage? Are you just using the free memory on the system? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VMware, * and SJphone ... newbie
I've read and tried a LOT of sample config's for sip.conf and extensions.conf and no matter what I do I get registration error's when trying to get SJphone registered to my * server. I have a XP VMware host with Redhat 9 / * as a guest. The SJphone is on the host XP trying to register with the guest Redhat/* server. Any suggestions? I think I'm just missing something stupid. thanks - A.G. Edwards & Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. -
Re: [Asterisk-Users] Ringing Delay
Chances are it's waiting to get the caller ID info (sent between the
first and the second ring)
On Wed, 2004-03-03 at 12:01, WipeOut wrote:
> Brian Mulligan wrote:
>
> >Sorry if this is a daft question but when a PSTN call comes in on my
> >X100P the console shows the following;
> >
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >-- Executing Dial("Zap/1-1", "Zap/2") in new stack
> >-- Called 2
> >-- Zap/2-1 is ringing
> >-- Zap/2-1 is ringing
> >WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
> >Didn't finish Caller-ID spill. Cancelling.-- Zap/2-1 is ringing
> >-- Zap/2-1 answered Zap/1-1
> >-- Attempting native bridge of Zap/1-1 and Zap/2-1
> >-- Hungup 'Zap/2-1'
> > == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
> >-- Hungup 'Zap/1-1'
> >
> >
> >AS indicated, the call is switched to the context [incoming] which is
> >configured as follows;
> >
> >[incoming]
> >exten=> s,1,Dial,Zap/2
> >
> >The extension rings but not until the incoming line has rung three
> >times. If I hang-up the external line before the extension is answered
> >then the extension continues to ring three more times. Clearly, if I
> >pick up the extension during this time then nobody is there!
> >
> >I would like the extension to ring immediately when the call comes in.
> >Despite my best efforts I cannot find a configuration element which
> >addresses it. Could it be my hardware just being too slow?
> >
> >Thanks
> >Brian
> >
> >
> >
> Brian,
>
> AFAIK the delay is caused by the way the X100P detects ringing.. If you
> are using digital lines (ISDN) then there is a signal to tell the device
> connected to the line that it is ringing but with an analog line this is
> not the case..
>
> So the X100P basically looks for the "swings" on the line that indicate
> that it is ringing, it goes through about 3 of them before it answers to
> avoid "phantom" calls which used to happen a lot and were very
> irritating especially in the middle of the night.. So what you are
> experiencing is normal..
>
> Later..
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/
BTEL Consulting
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
Mee too with the same problem Patrick Lidstone (Personal E-mail) said: >> From: Iain Stevenson <[EMAIL PROTECTED]> >> >> Anyone else seeing SIP registration requests rejected by FWD? >> I don't seem >> to be able to register any longer - even though my SIP config >> remains the >> same. >> >> Iain > > Yes, me too - for about the last week I'd guess. I'm guessing that it is > this that seems to have made my current asterisk installation really > unstable - the box locks up completely after a few hours (of failed > registrations?), failing silently with no errors recorded in the logs. > I've disabled FWD registration this morning, and time will tell if the > box is now more stable. > > Patrick > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best ATA 186 Firmware
Hi: Someone know wich is the best firmware for the ATA 186 with * Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs
If people on the list have ways to present to my customers ways to help me sell this product to my customer due to concerns about SCO. I want to hear it. I do agree that having general discussions about it is not what this list is meant for. Michael J. Mimbach II [EMAIL PROTECTED] WNOC / Mountain Wireless - Original Message - From: "Tilghman Lesher" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 11:03 AM Subject: Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs > On Wednesday 03 March 2004 11:47, Alex Lopez wrote: > > I will respond off list, to conserve bandwidth. > > Which you didn't do. > > > I feel that this is VERY ON-TOPIC. If this continues, we will be > > faced with customers shying away from Linux due to the whole FUD > > factor (fear, Uncertainty, and Doubt). > > Then post it to a generic Linux list. This list is about using > Asterisk specfically, not Linux generally. There are more than > enough lists out there for which such a post is on-topic. Please > take your non-Asterisk posts there. > > I don't think there's a single user on this list who cares about the > SCO issue who hadn't already heard this news elsewhere. In any > case, this list does not need to be the central ground for general > Linux news. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of SIP with an outgoing proxy?
Hey all, Doing a search of the mailing list archives turns up a couple requests for support of an outgoing SIP proxy, and the following wishlist request: http://bugs.digium.com/bug_view_page.php?bug_id=359 I'm trying to get a Vonage softphone account working, and it's mostly working, but it appears that I need the outbound proxy support for it to work. Basically, I can register and such on port 5061, but when I try to do an outgoing call it uses the URI: To: sip:[EMAIL PROTECTED]:5061 which results in a 403 or 404 error. When I make a call with the softphone and dump the traffic with Ethereal, the To is: To: sip:[EMAIL PROTECTED] which should default to port 5060. If anyone knows of another way to get Asterisk to copy this behavior, I'd be all ears. :) Oh, if I use port 5060 directly, I *can* make calls to other Vonage phones, but I get a 'Forbidden' when trying to dial non-Vonage numbers. I also can't register on port 5060. :( | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs
On Wednesday 03 March 2004 11:47, Alex Lopez wrote: > I will respond off list, to conserve bandwidth. Which you didn't do. > I feel that this is VERY ON-TOPIC. If this continues, we will be > faced with customers shying away from Linux due to the whole FUD > factor (fear, Uncertainty, and Doubt). Then post it to a generic Linux list. This list is about using Asterisk specfically, not Linux generally. There are more than enough lists out there for which such a post is on-topic. Please take your non-Asterisk posts there. I don't think there's a single user on this list who cares about the SCO issue who hadn't already heard this news elsewhere. In any case, this list does not need to be the central ground for general Linux news. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ringing Delay
Brian Mulligan wrote:
Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Dial("Zap/1-1", "Zap/2") in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
Didn't finish Caller-ID spill. Cancelling.-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
-- Hungup 'Zap/2-1'
== Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
AS indicated, the call is switched to the context [incoming] which is
configured as follows;
[incoming]
exten=> s,1,Dial,Zap/2
The extension rings but not until the incoming line has rung three
times. If I hang-up the external line before the extension is answered
then the extension continues to ring three more times. Clearly, if I
pick up the extension during this time then nobody is there!
I would like the extension to ring immediately when the call comes in.
Despite my best efforts I cannot find a configuration element which
addresses it. Could it be my hardware just being too slow?
Thanks
Brian
Brian,
AFAIK the delay is caused by the way the X100P detects ringing.. If you
are using digital lines (ISDN) then there is a signal to tell the device
connected to the line that it is ringing but with an analog line this is
not the case..
So the X100P basically looks for the "swings" on the line that indicate
that it is ringing, it goes through about 3 of them before it answers to
avoid "phantom" calls which used to happen a lot and were very
irritating especially in the middle of the night.. So what you are
experiencing is normal..
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs
I will respond off list, to conserve bandwidth. I feel that this is VERY ON-TOPIC. If this continues, we will be faced with customers shying away from Linux due to the whole FUD factor (fear, Uncertainty, and Doubt). We have all made a commitment be it financial or simply time; we all have an interest in this. Laws can protect but they can also smoother. For a clear example look at the PSTN telecom business in the US. It is a lawyer's dream!! I understand that we all have different views of laws and the way the affect us. However, having something or rather a company like SCO change the rules simply because they want the 'best for the shareholders' scares me. Message: 3 From: Tilghman Lesher <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SCO finds someone to pay!!! Date: Wed, 3 Mar 2004 10:56:04 -0600 Reply-To: [EMAIL PROTECTED] On Tuesday 02 March 2004 22:30, Alex Lopez wrote: > I don't believe this!! SCO got some one to pony up 7 figures!! Please don't post off-topic crap like this. I get enough of this on other lists. -Tilghman --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
Iain Stevenson wrote: Anyone else seeing SIP registration requests rejected by FWD? I don't seem to be able to register any longer - even though my SIP config remains the same. I wouldn't worry about it - FWD goes through phases of failed registrations. It's a very highly used service, and hence (imo) somewhat overloaded, especially since they did the free calls to the US/Canada over chrismas last year. Give it a few hours/days and it should spring back into life. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
- Original Message - From: Patrick Lidstone (Personal E-mail) <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 5:28 PM Subject: [Asterisk-Users] FWD registration faillures > > From: Iain Stevenson <[EMAIL PROTECTED]> > > > > Anyone else seeing SIP registration requests rejected by FWD? > > I don't seem > > to be able to register any longer - even though my SIP config > > remains the > > same. > > > > Iain > > Yes, me too - for about the last week I'd guess. I'm guessing that it is > this that seems to have made my current asterisk installation really > unstable - the box locks up completely after a few hours (of failed > registrations?), failing silently with no errors recorded in the logs. > I've disabled FWD registration this morning, and time will tell if the > box is now more stable. > > Patrick I been sufferring from this problem recently. I noticed whilst investigating another problem that the host replying to my register has changed from fwd.pulver.com to oldfwd.pulver.com. Still the same IP of course :-) Is there an upgrade going on. Steve B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing Delay
Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Dial("Zap/1-1", "Zap/2") in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
Didn't finish Caller-ID spill. Cancelling.-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
-- Hungup 'Zap/2-1'
== Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
AS indicated, the call is switched to the context [incoming] which is
configured as follows;
[incoming]
exten=> s,1,Dial,Zap/2
The extension rings but not until the incoming line has rung three
times. If I hang-up the external line before the extension is answered
then the extension continues to ring three more times. Clearly, if I
pick up the extension during this time then nobody is there!
I would like the extension to ring immediately when the call comes in.
Despite my best efforts I cannot find a configuration element which
addresses it. Could it be my hardware just being too slow?
Thanks
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD registration faillures
> From: Iain Stevenson <[EMAIL PROTECTED]> > > Anyone else seeing SIP registration requests rejected by FWD? > I don't seem > to be able to register any longer - even though my SIP config > remains the > same. > > Iain Yes, me too - for about the last week I'd guess. I'm guessing that it is this that seems to have made my current asterisk installation really unstable - the box locks up completely after a few hours (of failed registrations?), failing silently with no errors recorded in the logs. I've disabled FWD registration this morning, and time will tell if the box is now more stable. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
- Original Message - From: AstGrp <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 01, 2004 5:22 AM Subject: RE: [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk > I have this working, with not much work... > > SIP CONF > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; address to bind to > externip = ; Address that we're going to > put in SIP messages if we're behind a NAT > localnet = 10.100.254.0 ; Internal NETWORK address > localmask = 255.255.255.0 ; Internal netmask > context=default ; Default for incoming calls > ;srvlookup = yes; Enable SRV lookups on outbound calls > ;pedantic = yes ; Enable slow, pedantic checking for > Pingtel > ;tos=lowdelay > ;tos=184 > ;maxexpirey=3600; Max length of incoming registration we > allow > ;defaultexpirey=120 ; Default length of incoming/outoing > registration > ;notifymimetype=text/plain ; Allow overriding of mime type in > NOTIFY > ;videosupport=yes ; Turn on support for SIP video > disallow=all; Disallow all codecs > allow=ulaw ; Allow codecs in order of preference > allow=ilbc > allow=alaw > > > [travel] > type=friend > username=travel > secret= > host=dynamic > nat=yes > context=local > mailbox=4003 > > Ports in the Firewall > > Port 5060 UDP > Ports 16456 - 17456 UDP > > RTP Conf > > rtpstart=16456 > rtpend=17456 > > -gcc > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steve > Beaumont > Posted At: Sunday, February 29, 2004 4:12 PM > Posted To: Asterisk User Group > Conversation: [Asterisk-Users] Asterisk as a SIP server behind nat, > clients on the outside connecting to Asterisk > Subject: [Asterisk-Users] Asterisk as a SIP server behind nat, clients > on the outside connecting to Asterisk > > > On the wiki pages it suggests that clients on the outside of NAT can > connect to an Asterisk server behind nat. (option no 3). The note > suggests that this can work with port forwarding and some 'header > mangling magic'. > > I have the port forwarding configured however, when I try to connect an > external client through the firewall the client does not correctly > register. The REGISTER message is received, the server responds with > Status 100 trying, followed by Status 407 Proxy Authentication required. > This repeated several times. > > I guessing but could this be where the 'header mangling magic' is > required. ? Does anyone know how this magic can be applied. > > Many thanks > Steve Beaumont Thanks for the replies, but this has turned out to be a little more involved than it first appeared. I'm sorry to say I haven't really got to the bottom of it yet but it seems to be a problem with the way my router handles NAT/PAT. Unfortunately, I am unable to sniff the adsl side of my internet connection so it's proving a little difficult to pin down. I must admit I'm a little surprised that a fairly recent protocol like SIP is not more firewall /NAT/PAT friendly. Anyhow, less of the moaning. A general question that has been on mind for a while is the range of RTP ports used by SIP. What governs there allocation. Concurrent connections ? All the best Steve Beaumont ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
Robert Sprockeels wrote: Mar 3 17:57:30 NOTICE[147466]: chan_iax2.c:3209 authenticate: Asked to authenticate to 69.73.19.178 with an RSA key, but they don't allow RSA authentication I've found FWD to either work or not, it's been on and off for the last few days... which is why i turned to enum as a method of contacting other people without the problems of a congested service such as FWD... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
On Wed, 2004-03-03 at 17:44, Iain Stevenson wrote: > Anyone else seeing SIP registration requests rejected by FWD? I don't seem > to be able to register any longer - even though my SIP config remains the > same. Sorry for my previous post... it's my iaxtel registration that fails! My FWD registration succeeds. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD registration faillures
On Wed, 2004-03-03 at 17:44, Iain Stevenson wrote: > Anyone else seeing SIP registration requests rejected by FWD? I don't seem > to be able to register any longer - even though my SIP config remains the > same. Same here. Error message is: Mar 3 17:57:30 NOTICE[147466]: chan_iax2.c:3209 authenticate: Asked to authenticate to 69.73.19.178 with an RSA key, but they don't allow RSA authentication Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCO finds someone to pay!!!
On Tuesday 02 March 2004 22:30, Alex Lopez wrote: > I don't believe this!! SCO got some one to pony up 7 figures!! Please don't post off-topic crap like this. I get enough of this on other lists. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status Lights on Snom200 Phone Displaying the Status of PSTN Lines
Alright, this may seem like something relatively easy to do
but I must be missing something or had a neuron misfire. I am trying to
get
The Status lights on my Snom200 hardphones to display the
status of each one of my PSTN lines in my Asterisk server.
Current Config:
3 X100P cards
Asterisk CVS-02/25/04-18:06:52
5 Snom200 phones
I am currently using the following macro to dial out with my
extensions
[macro-stdexten]
exten =>
s,1,Dial(${ARG2},20,Ttr)
; Ring the interface, 20 seconds maximum
exten =>
s,2,Voicemail2(u${ARG1}) ; If unavailable,
send to voicemail w/ unavail announce
exten =>
s,3,Goto(default,s,1) ;
If they press #, return to start
exten =>
s,102,Voicemail2(b${ARG1}) ; If busy, send to voicemail
w/ busy announce
exten =>
s,103,Goto(default,s,1) ; If they press
#, return to start
My extension can then use this as follows:
exten => 202,1,Macro(stdexten,${USERVM},${USER})
I do not have an extension setup directly to dialout and I
am thinking that this might be part of my problem. I would also like to
know the status of certain extensions
To see who is on the phone and who is not by looking at the
soft key light on the Snom200. If anyone has any ideas, they would be
greatly appreciated.
Sincerely,
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines, IA 50309
Phone: (515)-244-6290
Cell: (720)-841-5802
Website: www.dstorage.com
E-Mail: [EMAIL PROTECTED]
[Asterisk-Users] NAT, Asterisk and SIP service provider (sipgate.de)
Hello Oliver,
okay, this was not easy and will make a long e-mail that I will also CC
to the list. I will answer in English because it is my native language.
I lived in Germany for 2.5 years and can speak German okay,
however I will spare you all of the declination failures that I make
on a regular basis.
I have an OpenBSD NAT'ting firewall allowing asterisk to talk to
sipgate.de with outgoing calls working nicely, incoming is untested
but should work.
sipgate.de is provides their services using SIP, and asterisk can be a
SIP client, you probably know this. SIP service providers setup their
systems to support "normal" SIP clients and you need to make you
asterisk and firewall (the "and firewall" bit is perhaps the most
important) appear to be a normal SIP client at the UDP port level.
SIP uses UDP port 5060 as its call setup/control port and some UDP
ports for its RTP media stream. The RTP media stream ports are set in
the asterisk control file rtp.conf.
I analyzed the traffic at the port level using xten's x-lite SIP
client talking to sipgate and discovered that the firewall setup is
very important. If you use NAT, standard procedure is to take outgoing
connections and translate them using some "random high port" as the source
port. so:
SIP Client <> NAT Firewall <-> sipgate.de
int ip : UDP 5060 NAT to: ext ip UDP 645035sipgate.de UDP 5060
The NAT firewall then keeps this config and expects to route info back
from sipgate to the internal SIP client on UDP port 645035. However
sipgate and the RFC think that SIP clients should accept info on UDP
port 5060 so it sends info back to (ext ip) UDP port 5060 and the
firewall may route this but it is not part of the same connection and
so it seems to get lost somehow.
What needs to be done is to tell the firewall to route all connections
on UDP 5060 out using UDP port 5060. in OpenBSD the pf.conf extries
look like this:
/etc/pf.conf:
# outgoing UDP port 5060 connections use source port 5060 on firewall
nat on $ext_if inet proto udp from any port = 5060 to any -> ($ext_if) port 5060
# incomming UDP port 5060 connections should go to my asterisk server
rdr pass on $ext_if proto udp from any to ($ext_if) port 5060 -> $voip_box
#RTP MEDIA STREAM redirect.
rdr pass on $ext_if proto udp from any to any port :20001 -> $voip_box port
:20001
When this works, and keep in mind that this is for OpenBSD (I am not
sure if linux can do this), then asterisk setup is as follows:
/etc/asterisk/sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
externip = xxx.sjwilliamson.ca
localnet = 192.168.1.0 ; Internal NETWORK address
localmask = 255.255.255.0 ; Internal netmask
nat = yes
register => 8007163:[EMAIL PROTECTED]/8007163
[sipgate]
secret=xxx
username=8007163
fromuser=8007163
fromdomain=sipgate.net
type=friend
host=sipgate.de
nat=yes
;qualify=yes
dtmfmode=rfc2833
canreinvite=no
context=in-sipgate
/etc/asterisk/rtp.conf - this is stock
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=1
rtpend=2
/etc/asterisk/extensions.conf
;outgoing sipgate calls
[sipgatede]
exten => _0049.,1,SetCallerID(4921158007163)
exten => _0049.,2,SetCIDName(Scott Williamson)
exten => _0049.,3,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30)
exten => _0049.,4,Playback(the-party-you-are-calling)
exten => _0049.,5,Playback(is-curntly-unavail)
exten => _0049.,6,Hangup
exten => _0049.,104,Playback(the-party-you-are-calling)
exten => _0049.,105,Playback(is-curntly-busy)
exten => _0049.,106,Wait,3
exten => _0049.,107,Hangup
;incomming sipgate calls
[in-sipgate]
exten => 8007163,1,Macro(stdexten,1234,${PHONE1})
Incomming calls in the context [in-sipgate] need to have an extension
that is the same as your sipgate number. And you need to register with
this also.
Good luck, and remember that in this case the firewall config is the
most important, second is the extension / sipgate number in the
registration and in the context [in-sipgate].
Also, "show sip registry" at the asterisk console will show if you
have registered with sipgate. They seem to go offline sometimes, and I
do not know why. I consider this to be normal, as this happens to
other SIP accounts that I have.
Scott Williamson
P.S. Maybe you can try calling me over sipgate @ +49 211 58 00 71 63 to test and
see if incoming calls work.
--
Best regards,
Scottmailto:[EMAIL PROTECTED]
---
|Toronto | +1 416 xxx | PSTN |
|-|---|-|
| Düsseldorf | +49 211 58 00 71 63 | International |
| London | +44 20 71 27 63 82 | PSTN & ENUM|
|-|---|-|
|FWD | 25 39 84 | VOIP |
| iaxTel | 1 700 839 8593 |
[Asterisk-Users] FWD registration faillures
Anyone else seeing SIP registration requests rejected by FWD? I don't seem to be able to register any longer - even though my SIP config remains the same. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriks & ALSA???
When I upgraded from OSS to the ALSA soundsystem, I could no longer get noice from the microphone. I changed my modules.conf to; load => chan_alsa.so noload => chan_oss.so And my alsa.conf to; input_device=hw:0,24 output_device=default or input_device=hw:0,0 output_device=default or input_device=hw:24,0 output_device=default or ;input_device=hw:0,24 ;output_device=default or input_device=default output_device=default but none of the configurations worked. I changed all channels with alsamixer and gnome-alsamixer, but no result. Is there somebody who knows what's going wrong Anton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfers from SIP
Sorry for such a basic question, but Googling and wiki searches haven't
lead me anywhere.
Can a called phone transfer a call to another number?
In detail, I have ISDN/BRI via chan_capi -> * -> SIP to some xlite
workstations.
All my dial() strings have tT on the end of them, typical is:
exten => _${ISDN}${EXT_DAVE},1,wait(1)
exten => _${ISDN}${EXT_DAVE},2,wait(1)
exten => _${ISDN}${EXT_DAVE},3,wait(1)
exten => _${ISDN}${EXT_DAVE},4,SetCallerID(${CALLERIDNUM})
exten => _${ISDN}${EXT_DAVE},5,\
dial(Console/${RINGTONE_DAVE}&sip/${EXT_DAVE}&CAPI/264555:b${MOB_DAVE},${RINGTIME},,tT)
exten => _${ISDN}${EXT_DAVE},6,VoiceMail(u1${EXT_DAVE})
exten => _${ISDN}${EXT_DAVE},7,hangup
exten => _${ISDN}${EXT_DAVE},106,VoiceMail(b1${EXT_DAVE})
exten => _${ISDN}${EXT_DAVE},107,hangup
The Console call is just used to 'broadcast' a ringtone round the office.
What have I missed? I've tried xlite with forced inband DTMF and unforced,
I've tried to trace the source code of transfer actions, I'm lost!
Dave Kitchen
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
Email back from ipblue: Thank you for your inquiry. At the present time we work under SCCP or H.323 protocols. We are developing a SIP phone which will be available sometime in Q2. Regards, Andrew Schecter Vice President of Sales IP blue Software Solutions 15 East 26th Street New York, NY 10010 212.485.1225 (tel) 212.485.1380 (fax) [EMAIL PROTECTED] -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Detection
Does the fax detection only work with an X100P or Zaptel card? Will asterisk auto detect a fax on an incoming SIP call from a Background menu? Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX image for SNOM 200?
Earlier on I read that there is an IAX image for the SNOM 200. Is this true? Does it work? Where might I get this? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] KPN BRI
> The software configuration depends (of course) on your hardware > configuration. In this situation, what kind of hardware do you plan to > use to connect your asstricks box to KPN? You are best of going > with a capi capable isdn card, and using chan_capi (see > http://www.junghanns.net/asterisk/page1.html), your > alternative would be to use chan_modem together with isdn4linux. > I've done both, so my is based on personal experience. I have 2 Eicon Diva cards which I am using chan_capi. I have chan_capi installed and configured and it detects the ports ok. I have the lines plugged in but when I dial the number associated with the line does not get picked up and I get a non-existant number tone. When I plug in a standard isdn telephone into the line it all works ok. I think I am using the wrong kind of signalling. I have found out that kpn use e164 as the signalling but I cannot find anywhere to configure this. Thanks for any advice you can give. Regards Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Troubles with Galaxyvoice
Try [phone]:[password]:[EMAIL PROTECTED] this is what I had to use for one of the providers I use (iconnecthere) Martin On Wednesday 03 Mar 2004 3:00 pm, Mark Phillips wrote: > Folks, > > I have subscribed to galaxyvoice for $20 and so far everything is fine as > long as I use the Grandstream phone I bought from them. > > What I want to do is use *. They claim that they support SIP and that I > can use any SIP client with them. However, their tech support sucks and > I'm unable to register with them. I have to admit though that they do > actually have humans on the end of the phone which is a novelty for an > ITSP. > > OK hears my problem. They gave me 3 bits of info; a phone number, username > and password. I've tried registering with user:[EMAIL PROTECTED], > phone#:[EMAIL PROTECTED] and phone#:[EMAIL PROTECTED] and am unabel to get > registered. > > I am also unable to give them calls. They require the g.729 codec which I > bought from digium (not bad for $10) but I'm still unable to complete a > call with them. I suspect its for the same reason as not being able to > register. > > Any ideas would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not ringing.
RTP stream not passed through the * server in case of SIP(Sipura) > H323(Cisco) traffic (Grand Stream was doing OK so we suspected SIPURA). After SIPURA firmware upgrade (What ever latest) started working correctly. No confirmed reason but Firmware upgrade did the trick. - SamW -Original Message- From: Andres [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 11:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura 2000 not ringing. Hi Sam, Can you elaborate on the rtp issues you saw? I am interested in hearing about them. Regards, Andres. - Original Message - From: "SamW" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 3:19 PM Subject: RE: [Asterisk-Users] Sipura 2000 not ringing. > > Did you try to upgrade the firmware?, some issues we saw with rtp > stream, went away after a firmware upgrade. > > http://www.sipura.com > > -SamW > > > > -Original Message- > From: Mark Messmore, Technical Support, University Telcom Inc. > [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 02, 2004 1:59 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Sipura 2000 not ringing. > > I was just wondering if anyone has had this situation...or one similar > to it. > > I've got a Sipura SPA 2000. After hooking it up and configuring it with > my * box, it has worked well. From both lines we are able to dial out > at any point in time. However after a few minutes (5-10 usually) the > Sipura will stop sending a ringer signal to the phones. We can still > dial, * shows that it is ringing those SIP clients, and both lines are > still shown as being "registered". Everything will work fine for the > first 5-10 minutes after being "rebooted"...however after that there is > silence. Even if I dial the line from my cell phone and pick up the > telephone that I am dialing...there is dial-tone...therefore it seems > that the Sipura is just not passing those incoming calls after a short > time period. > > If you have any idea why this is happening I'd sure appreciate hearing > your thoughts. Thanks > > Mark > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Troubles with Galaxyvoice
Folks, I have subscribed to galaxyvoice for $20 and so far everything is fine as long as I use the Grandstream phone I bought from them. What I want to do is use *. They claim that they support SIP and that I can use any SIP client with them. However, their tech support sucks and I'm unable to register with them. I have to admit though that they do actually have humans on the end of the phone which is a novelty for an ITSP. OK hears my problem. They gave me 3 bits of info; a phone number, username and password. I've tried registering with user:[EMAIL PROTECTED], phone#:[EMAIL PROTECTED] and phone#:[EMAIL PROTECTED] and am unabel to get registered. I am also unable to give them calls. They require the g.729 codec which I bought from digium (not bad for $10) but I'm still unable to complete a call with them. I suspect its for the same reason as not being able to register. Any ideas would be greatly appreciated. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Size of PC for conferencing?
Hi Ariel- I wonder if you could please expand on that a little? What was your configuration for conferences when you had the problem (how big were the conferences, were there errors in the /var/log/asterisk/messages file, etc) I do have problems in a setup with two TE410P's, although my environment is IVR oriented, and the problems I'm experiencing seem to be related to driver deficiencies in handling PRI buffering errors. Also, why do you think the E400P would work better - is it just because its only one E1 instead of four? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Email: scott "at" evtmedia.com URL:www.evtmedia.com >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of >Ariel Batista >Sent: Wednesday, March 03, 2004 2:38 PM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Size of PC for conferencing? > >Tony Mountifield wrote: >> Can anyone advise from experience what size of PC would be needed >> to support two TE405P 4xE1 cards to provide conference bridging >> for up to 20 concurrent conferences of 10 participants each? > >>From working with 2 TE410P in one PC I can tell you that it will not >work. And from further testing on system we found that if you >have more >then 2 E1 or T1's in a system even with a heavy duty dual Xeon the >system will crash after about 40 conferences. > >So you will need many servers and would recommend you going to >the E100P >card instead and more servers. > >> All the participants would be on the E1 trunks, not VoIP. >> >> Thanks in advance, >> Tony > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel on Debian
On Wed, 2004-03-03 at 00:34, Hermann Wecke wrote: > On Thu, 5 Feb 2004, Tim Sailer wrote: > > Does anyone have the zaptel modules built for Debian 2.4.24 kernel? > > Someone here is running * on debian? I have * running on Debian stable. I back-ported the zaptel and asterisk packages from testing. I'm currently evaluating * for my uses and so this is not a production system. My back-ported packages are also not complete although they are functional for my uses so far. > Recompiled kernel to latest (?) 2.4.24 > uname -a > Linux mail 2.4.24 #5 SMP Tue Mar 2 17:31:13 BRT 2004 i686 GNU/Linux stegosaurus:/etc/asterisk# uname -a Linux stegosaurus 2.4.18-1-686 #1 Mon Jan 5 19:32:08 UTC 2004 i686 unknown do you have a hardware problem? the zaptel hardware "just worked" for me. -joe -- Innovation Software Group, LLC - http://www.innovationsw.com Custom Internet and Computer Solutions Linux, UNIX, Java Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM[RBL] Re: [Asterisk-Users] Calls not hanging up.
Thanks, I figured it was the telcos problem but I appreciate knowing it is. I wouldn't think that Telus (here in Canada) would have equipment that old hanging around but I guess who knows. Thanks for your help, Darren Wiebe [EMAIL PROTECTED] John Fraizer wrote: Darren Wiebe wrote: The complaint I'm getting from a few people is that when they hang up their phones, they still cannot get dialtone for a while. Two people said last night that even 20 seconds after they hung up their phones, when they picked up again, they still did not have a dial tone. I'm not sure when it came back. For most people it works fine. Any suggestions? I don't think it is their phones because it worked fine for both of them other times. Then again, I don't know what it could be besides their phones. It is circuit supervision. Sounds like they are on some OLD telco switches on their end. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does usb-ohci work for ztdummy?
On Wed, 2004-03-03 at 14:57, Zen Kato wrote: > Hi, > > One of my CPU boards is MS-5169(ATX AL9 mainboard). The chipset is > Aladdin M1531/M1543. The USB is 'usb-ohci'. ztdummy.c and ztdummy.h > uses 'usb-uhci', so I changed from 'uhci' to 'ohci' in ztdummy.c and > ztdummy.h. and tried "/sbin/modprobe ztdummy", never succeeded. > > Is it impossible to use 'usb-ohci' instead of 'usb-uhci' for ztdummy? Yes it needs usb-uhci. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
