Re: [Asterisk-Users] ATA's
hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs Excuse me I have just bought a PAP2 ,, is it true that only one g729, one of the Damn things Cisco had in the ATA186? at the same time. DAMN , its just a Sipura inside I really dont know why is this, offer a 1 port version or a optional second port g729 is really a pain this. regards HA On Tue, 15 Feb 2005 08:52:21 +0100, Nicolas Bougues [EMAIL PROTECTED] wrote: On Mon, Feb 14, 2005 at 10:47:23PM +0900, Hermann Wecke wrote: Matthew Boehm wrote: [...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... Back in december, the Uniden was supposed to do 2xG729 at a later time. Not sure if the current firmware allows it. BTW, I've been fairly disappointed with Uniden firmware and their release cycle : their hardware is great, but they take months to release new firmwares, even when phone crashing bugs are discovered. If you want 2xG.729 now, working reliably, for under $90, you can't go wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a bridge mode, where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk restart alone
Hello, I have an Asterisk server. When I connect to the console (asterisk -r) and I want to see the time that the server has been connected (CLI show uptime) I noticed that Asterisk restarts alone. Why? Any clue will be apreciated. Best Regards, Thank´s.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? Thanks, -- Lord, protect me from your followers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip phones how to dial a # sign?
Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote: Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? It could quite possibly be related to the log files the polycom uploads to the FTP server. I found this quite a pain, so disabled all the logging in the config files. If that isn't it, then you will need to find out *what* activity the polycom is doing, hint tcpdump -tn -A -s 16384 port 21 might help or else see your ftp server log files. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, Feb 15, 2005 at 07:56:10PM +1100, Adam Goryachev wrote: On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote: Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? It could quite possibly be related to the log files the polycom uploads to the FTP server. I found this quite a pain, so disabled all the logging in the config files. If that isn't it, then you will need to find out *what* activity the polycom is doing, hint tcpdump -tn -A -s 16384 port 21 might help or else see your ftp server log files. You are right, this activity is related to logging. After consulting the admin manual I am unsure as to what settings related to logging are safe to change (some are marked as don't modify without consulting Polycom). Do you remember which settings you changed to disable logging? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote: You are right, this activity is related to logging. After consulting the admin manual I am unsure as to what settings related to logging are safe to change (some are marked as don't modify without consulting Polycom). Do you remember which settings you changed to disable logging? I changed the settings that it told me not to... Basically, I think I changed the various log levels to don't log anything... Hope that helps, if not, let me know and I can send you the polycom file off-list... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which IP phone to use in Australia
Hi guys, I haven't had the opportunity to play with any Polycom products, although they will probably be the best IP phone available. I have used the Grandstream BT-101/102, the HOP-1003 (upgraded 1002) and Zyxel telephone adapters. My recommendation out of the tried ones would be the Grandstream BT-102 where the phone is on a closed network. My problem with the HOP is that there is no transfer button, this can be worked around with key sequences (*2 for attended transfer, # for unattended transfer) and (if I got it working) the flash key but they aren't optimal in an office setup where un-techno people want to just transfer a call with a button and the overall receiver volume seems to be lower (definitely than the analogue adapter). The HOP does have one advantage: it only beeps once when you get an incoming call along with a message on the display unlike the BT-101/102. The Zyxel equipment works great for us and it supports callerID (only the number) but it does mean the cost of an analogue phone. If you have an issue with network cabling, the Zyxel is good because you can run two analogue phones from one network cable, and there is a hub built into the back which allows inline connection to a PC. This having been said, because they are analogue, you need to use the *2 and # keys for transfer etc. I would be interested in trying (and have been offered but haven't had the time) a new Sipura telephone, it includes two line indicators. Whilst this isn't ideal for a receptionist (I think Snom would be the only option for a receptionist with the additional indicator panel) it would be nice for a general office worker who needs one direct line and one general ring group / reception button. They can be expanded to four lines with some sort of software upgrade. One thing to be aware of - where you will be setting up standalone offices (i.e. one person at home behind DSL) you should consider the firewalling etc but seeing as the original question came from the experts - they should be able to sort it out! Our experience is that some hardphones will even have troubles with specific firewalls, yet they will work quite happily with a Billion style product (sorry to bring that up). Hope this helps. Kind Regards Stuart On Tuesday, Feb 15, 2005, at 15:33 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-15 at 18:05, Adam Goryachev wrote: On Tue, 2005-02-15 at 17:54 +1100, Howard Lowndes wrote: On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote: Personally, I quite like the polycom phones such as the IP300 and IP600 I've never really bothered with the IP500... There are a few issues I have with them though, the main one being that I can't disable call waiting on the phone. There are workarounds for this though (in asterisk dialplan). ...which is something to be said for the HOP 1002 - you can disable call waiting. Have you actually used the polycom phones? If so, how do they compare to the HOP 1002, or, would you call the polycom IP600 and HOP 1002 exactly equivalent in all respects except for the call waiting factor? Unfortunately I have never used, or even seen the polycom phones, so I cannot comment on the comparison. I do know that the HOP 1002 serve my purpose and are quite robust. There was a date issue with the software pre v1.41.007 and I have found out how to get a brand name to display on the screen. I have also discovered that, under SIP at least, the phone will only display the caller ID number and not the caller ID name, though that latter is not often sent anyway except for calls from mobiles as MOBILE. Basically they are very robust, almost brick shithouse robust. :) The online manual is about 47 pages of Chinglish which is an Alexander (downer). (Oz joke there for all you yanks) The only down side that I can see is that the 2 port hubbing is only 10 mbps which shouldn't really be a problem for most users who connect their PC in line, but could be a real bummer for the power user PHBs who want to do gaming. I've not seen/used the HOP 1002, I just find it hard to accept that it would be as good as the polycom IP600 phones Note: I would be *pleasantly* surprised if you say it is as good! Regards, Adam -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4211a62d449061270431442! Stuart Elvish Business Development Manager TNet.com.au - Becoming Australia's Favourite Internet
[Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?
Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi channels to route calls ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID on PRI
On 15 Feb 2005, at 05:44, Rod Bacon wrote: Some more info on my problem that someone may be able to explain. The debug information (shown below), lists the LENGTH of the CallerID string as 14 characters, even though I'm only sending 10. I belive that this is the problem. My telco's equipment is looking for 10 characters only. Any ideas where these extra 4 characters are coming from? Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '0386172169' I think the 'extra' bytes are part of the data structure, and perfectly normal. I'm basing this remark on a very quick look at the source code. in dump_called_party_number() in llibpri/q931.c it says: q931_get_number(cnum, sizeof(cnum), ie-data + 1, len - 3); Which looks like the first byte and the last 3 bytes are protocol surrounding the actual number. My best advice is to call your PTT and ask them how many digits they expect you to send, I am guessing they only expect the last 2, but only they know for sure. If I get the time I'll do a debug on my E1 PRI span later today and send you the results. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?
On Tue, 15 Feb 2005 10:45:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi channels to route calls ? capiECT is probably what you are after. Have a look at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme Thanks in advance, regards, Rob. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, Feb 15, 2005 at 08:26:42PM +1100, Adam Goryachev wrote: On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote: You are right, this activity is related to logging. After consulting the admin manual I am unsure as to what settings related to logging are safe to change (some are marked as don't modify without consulting Polycom). Do you remember which settings you changed to disable logging? I changed the settings that it told me not to... Basically, I think I changed the various log levels to don't log anything... Hope that helps, if not, let me know and I can send you the polycom file off-list... OK, I changed only the following settings: log.render.realtime=0 log.render.stdout=0 log.render.file=0 and now proftpd is quiet. Thanks again, cheers, -- If you're not having fun right now, you're wasting your time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] prblem in compileing asterisk-prepaid
Hello Any one using asterisk-prepaid with mysql. i want asteirsk-prepaid for fedora core 2. i have installed mysql-devel. but after that i am unable to compile the asterisk-prepaid it is giving me error for libmysqlclient. i already have this library in my /usr/lib/mysql. i am using asterisk-CVS. Here is the error given when i try to compile asterisk-prepaid. [EMAIL PROTECTED] asterisk-prepaid]# make make -C apps make[1]: Entering directory `/asterisk-prepaid/apps' gcc -shared -Xlinker -x -o app_prepaid_auth_pin.so app_prepaid_auth_pin.o -lmysqlclient /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make[1]: *** [app_prepaid_auth_pin.so] Error 1 make[1]: Leaving directory `/asterisk-prepaid/apps' make: *** [all] Error 2 __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?
Robert Rozman wrote: I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. IIRC you can't do this. You must connect your ISDN PBX to a HFC card and route it thru there. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk@home in production env
Hi there I just wanted to know what the difference between [EMAIL PROTECTED] and manually built boxes actually is ?? What makes [EMAIL PROTECTED] a home system ? Is it not a good idea to run [EMAIL PROTECTED] then modify/tweak it to use in a production environment ??, if so why not, would somebody be able to explain this to me, it seems to have a hobbiest tag associated with it, and i just wanted to know the difference cheers Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with SIP Registration at PSTN Provider
Hi together, I have a asterisk running on a Debian testing system running flawlessly at least after starting the asterisk. The Server its running on has a fixed IP, no NAT, whatsoever and is reachable all the time. The Firewall has holes on port 5060 and for the RTP-range that asterisk is configured on. In my sip.conf I have a few register=lines and I can receive calls over those accounts. However after a few hours, days whatsoever asterisk is still thinking that those registrations are valid(CLI) but my provider (sipgate.de) sees the asterisk as offline and hence does not forward calls to my asterisk box. I tried playing with the *expirey values in the sip.conf but I have no clue how to test it properly cause it can work fine for days and then stop working ... Sipgate is running SER btw. Is that a known problem ? Has anyone a solution other than restarting asterisk every hour or so ? regards, Magnus Jungsbluth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone / lipz4
hi It looks interesting, but it is documented to support only old RedHat versions and they don't release source to let me recompile. I am not a big RedHat fan, but if I have to use it on the desktop, I would want something newer than RedHat 9. If you can tell me you are using it with a newer distro, that would help. Lipz4 works fine on my up-to-date Debian Unstable. One nice looking softphone is SFLphone (http://www.sflphone.org). I personally have best experiences with Kphone and Xlite beta. rgrds, -- Klemens Kasemaa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID on PRI
On Tue, 15 Feb 2005, tim panton wrote: My best advice is to call your PTT and ask them how many digits they expect you to send, I am guessing they only expect the last 2, but only they know for sure. Also ask them if they require a specific Type Of Number for the outgoing callerid. (Configure that with the prilocaldialplan option). Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI not installed
I own a ME600 EPIA Mini-ITX main board with the latest Debian distro (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, isdnactivecards installed. I have a QuadBRI module by Junghanns with bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, but I have some strange behaviour. All modules seems to be correctly installed and actives, but on /dev I find only capi20. Anyway, starting Asterisk, I recevive a 'CAPI not installed!' error on chan_capi load and I can't find why. Anyone has some idea? Note: Asterisk without the QuadBRI module and chan_capi is working well, but I have compiled it with explicit PROC=i386, because 'uname -m' returns i686, but the VIA processor does not support some of 686 instructions that the Asterisk executable uses. # lsmod | grep capi capidrv297480 isdn1282041capidrv capi177280 capifs60242capi kernelcapi466246c4,blpci,bldma,bl,capidrv,capi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed
On 11:52, Tue 15 Feb 05, A. Peverelli wrote: I own a ME600 EPIA Mini-ITX main board with the latest Debian distro (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, isdnactivecards installed. I have a QuadBRI module by Junghanns with bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, but I have some strange behaviour. All modules seems to be correctly installed and actives, but on /dev I find only capi20. Anyway, starting Asterisk, I recevive a 'CAPI not installed!' error on chan_capi load and I can't find why. Anyone has some idea? Note: Asterisk without the QuadBRI module and chan_capi is working well, but I have compiled it with explicit PROC=i386, because 'uname -m' returns i686, but the VIA processor does not support some of 686 instructions that the Asterisk executable uses. Are you running asterisk as user asterisk ? If so, you need to add this user to the dialout group. Otherwise it won't have access to the modem. hope this helps. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed
Are you running asterisk as user asterisk ? If so, you need to add this user to the dialout group. Otherwise it won't have access to the modem. hope this helps. I'm running asterisk with user 'root'. Asterisk user is in the dialout group and I try to start asterisk as user asterisk, with the same result. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question regarding SER/Asterisk functionality
Hi all, I'm currently looking for a VoIP platform to support the following features: Caller ID Call Waiting with caller ID Call Hold/Retrieve Three-way conference Calling Line Identity Presentation Call back last missed call Last called number redial User line locking/Call Barring (all current levels) Itemised bill Call Forward Call Forward on No Reply Call Forward on Busy Call Forward Unconditional VoiceMail - Message Waiting Indicator - Call back option Number Portability Secret Number Legal intercept Emergency Number Routing Anonymous Caller Rejection Directory Service update Feature code activation I have looked into both SER and Asterisk, and found out that SER should be the most appropriate platform since most of the users will use a SIP adapter/endpoint and be located on the Internet (like most of the VoIP services these days). So my question to the list is to gather some experiences and possible remarks/comments on which direction I should go :D Any help would be gladly appreciated :-) -- Geir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] solid-state asterisk pbx?
I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Modern flash devices (usb, compactflash) have builtin wear leveling management and will last longer than you think: http://www.sandisk.com/pdf/oem/WPaperWearLevelv1.0.pdf Use ramdisk to store temporary files and flash to store permanent pbx configuration data, voicemail etc. Done right, one could literally have a pbx on a stick. Eg a 256mb, 512mb or 1gb sandisk usb2.0 dongle. Anyone done something like this yet? -Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed
A. Peverelli wrote: I own a ME600 EPIA Mini-ITX main board with the latest Debian distro (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, isdnactivecards installed. I have a QuadBRI module by Junghanns with bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, but I have some strange behaviour. All modules seems to be correctly installed and actives, but on /dev I find only capi20. Anyway, starting Asterisk, I recevive a 'CAPI not installed!' error on chan_capi load and I can't find why. Anyone has some idea? quadBRI CAPI!!! The quadbri cards do not use/support CAPI. If you don't have another CAPI capable device in your system you can't/shouldn't use CAPI (I guess you could use CAPI via mISDN, but what is the point?) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and asterisk
Hello, Chan_capi can be used by a billion pci card S0? So i can fax througt it. Thank´s Em Fri, 11 Feb 2005 14:58:31 +0100 Stefan Gofferje [EMAIL PROTECTED] escreveu: Anabela Abreu schrieb: Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 AFAIK, chan_capi is for FritzCards with original AVM capi4linux only. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hello all, I have an asterisk 1.0.3 stable instaled on a box. All works fine with this machine, but the only problem i get is that suddenly the machine hangs up all the establised calls and we have to call again. This problem occurs twice a day and i don not know how to debug it. I read carefully the logs placed in /var/log/asterisk, but I can not find the reason for this hangs. I have to say that when this occurs, sometimes asterisk restart, if i make a "show uptime", I can see that asterisk has recently restart, but other times when the same occurs, I can see that asterisk is running seven hours ago, but our calls was hanged up too. I have to say that automaticaly (By own script) asterisk restart every night How could i debug that? Does your Asterisk hang suddenly all the establised calls? Do you know any command that help me finding the problem? Thanks for your time. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] solid-state asterisk pbx?
On Tue, 15 Feb 2005, quoth [EMAIL PROTECTED]: I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Anyone done something like this yet? Yes, I installed asterisk (Debian packages) on pebble[0] linux on a flash drive in a VIA Eden based system. This was one of those 800MB laptop-ide-emulating[1] flash drives, but the full install was 127MB so you could easily install it on a 256MB usb stick or similar. It's useful running asterisk on a read-only distribution like that, since it is resilient to people powering the system on and off without shutting it down first. - Matt [0] http://www.nycwireless.net/pebble/ [1] as in, it looks just like a laptop HDD, but is solid state internally. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk qualified
Good day all Is there any time of VOIP/SIP/asterisk qualifications or certificates? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] make of asterisk doesn't do anything...
I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, make just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking down a huge amount of processor time. I did a make clean before the make, but that didn't help anything. It is a slow machine, but I let it run for like 15m and it hasn't produced the first bit of output. Anyone run into this? Thanks for any advice... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] Anyone that knows this ATA?
hi the norwegian company nextgentel uses custom ATAs. does anyone know these by view? http://www.nextgentel.no/ressurser/brukerveiledninger/NextPhone.pdf thanks roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] solid-state asterisk pbx?
http://lists.digium.com/pipermail/asterisk-users/2004-March/038463.html follow the thread.. should give you some info - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 15, 2005 2:15 PM Subject: Re: [Asterisk-Users] solid-state asterisk pbx? On Tue, 15 Feb 2005, quoth [EMAIL PROTECTED]: I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Anyone done something like this yet? Yes, I installed asterisk (Debian packages) on pebble[0] linux on a flash drive in a VIA Eden based system. This was one of those 800MB laptop-ide-emulating[1] flash drives, but the full install was 127MB so you could easily install it on a 256MB usb stick or similar. It's useful running asterisk on a read-only distribution like that, since it is resilient to people powering the system on and off without shutting it down first. - Matt [0] http://www.nycwireless.net/pebble/ [1] as in, it looks just like a laptop HDD, but is solid state internally. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hangs the establised calls
Hello all, I have an asterisk 1.0.3 stable instaled on a box. All works fine with this machine, but the only problem i get is that suddenly the machine hangs up all the establised calls and we have to call again. This problem occurs twice a day and i don not know how to debug it. I read carefully the logs placed in /var/log/asterisk, but I can not find the reason for this hangs. I have to say that when this occurs, sometimes asterisk restart, if i make a "show uptime", I can see that asterisk has recently restart, but other times when the same occurs, I can see that asterisk is running seven hours ago, but our calls was hanged up too. I have to say that automaticaly (By own script) asterisk restart every night How could i debug that? Does your Asterisk hang suddenly all the establised calls? Do you know any command that help me finding the problem? Thanks for your time. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 4xHFC-s cards vs 1 quadbri HFC-4S card ?
Hi, I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120 EUR) and using 1 QuadBRI card (approx. 700 EUR) ? What makes such difference ? Is it possible to do first configuration ? With what drivers ? Is it stable ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System command causes core dump Warning: Newbie help :)
With the following program: #!/bin/sh # mailfax: program to email received fax as pdf FAXFILE=$1 RECIPIENT=$2 FAXSENDER=$3 FAXID=`basename $1|cut -d . -f1,2`.pdf FAXTXT=`basename $1|cut -d . -f1,2`.txt tiff2pdf $FAXFILE $FAXID sendfax.pl $FAXID $RECIPIENT $FAXSENDER $FAXFILE #end of program If I execute the following from the command line: mailfax /var/spool/asterisk/fax/1108470308.13060.tif [EMAIL PROTECTED] 017893 everything works just fine, and the .pdf comes through as an email attachment If I execute the following from the dialplan (* is CVS-HEAD 02/02/05) exten = 666,1,System(mailfax /var/spool/asterisk/fax/1108470308.13060.tif [EMAIL PROTECTED] 017893) then the tiff2pdf program generates a core dump. Why would this be the case ? What can I start looking for ? Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] solid-state asterisk pbx?
http://www.voip-info.org/wiki-Asterisk+Embedded+Systems - Original Message - From: [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, February 15, 2005 1:44 PM Subject: [Asterisk-Users] solid-state asterisk pbx? I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Modern flash devices (usb, compactflash) have builtin wear leveling management and will last longer than you think: http://www.sandisk.com/pdf/oem/WPaperWearLevelv1.0.pdf Use ramdisk to store temporary files and flash to store permanent pbx configuration data, voicemail etc. Done right, one could literally have a pbx on a stick. Eg a 256mb, 512mb or 1gb sandisk usb2.0 dongle. Anyone done something like this yet? -Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] solid-state asterisk pbx?
Hi Dan, I've been investigating the same thing. Try to Google for Asterisk+Soekris, Soekris is the company (http://www.soekris.com) that makes cute little 586 class fan-less single board computers that run both Linux and FreeBSD ... Good luck, Hans -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, February 15, 2005 12:45 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] solid-state asterisk pbx? I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Modern flash devices (usb, compactflash) have builtin wear leveling management and will last longer than you think: http://www.sandisk.com/pdf/oem/WPaperWearLevelv1.0.pdf Use ramdisk to store temporary files and flash to store permanent pbx configuration data, voicemail etc. Done right, one could literally have a pbx on a stick. Eg a 256mb, 512mb or 1gb sandisk usb2.0 dongle. Anyone done something like this yet? -Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Call recognition (call id)
Hi, Somebody already made call recognition with database access? Depending of call's number, it access a database looking for that number. Where can i find something about this? Thanks in advance Pablo Fernandes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make of asterisk doesn't do anything...
Michael, Someone may know a simple fix. If not, can you please install the 'strace' program, then run: strace -f -o /tmp/strace.out make This will run make, and log any system calls it makes to /tmp/strace.out. When it hangs, take a look in that file. It may have stopped on one system call, such as select() or poll(), or it may be scrolling endlessly, repeating the same system calls over and over again. If grep is using a lot of processor time, it's probably the latter. Either way, please paste the last 20 or lines into an email, and post it to this mailing list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Michael George wrote: I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, make just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking down a huge amount of processor time. I did a make clean before the make, but that didn't help anything. It is a slow machine, but I let it run for like 15m and it hasn't produced the first bit of output. Anyone run into this? Thanks for any advice... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
I have had this same problem. The only way I know is to disable transfers in asterisk. You can still use the transfer control in your SIP device. Of course this does not work with call parking. I would be very interested in a solution that does not require disabling of transfers in asterisk as well. Pedro On Tue, 15 Feb 2005 09:52:56 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail to detect DTMF over direct ISDN pri link
Title: Fail to detect DTMF over direct ISDN pri link Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the monitor command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. Sylvain Gagnon Speech Technology Integrator BCE Elix Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question regarding SER/Asterisk functionality
Geir, Many of your items, such as Voicemail, are not supported by SER directly. It sounds, at least at this very early stage, as though you'd be better off with Asterisk as it supports all of these features, though perhaps with some development work. If need be, SER could front it for call routing and scalability. My company, Integrics Ltd, can offer you formal advice on this, and can also provide installation and development services for both Asterisk and SER. If you'd like more details, drop me an email off list, or phone me on the number below. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Geir O. Høgberg wrote: Hi all, I'm currently looking for a VoIP platform to support the following features: Caller ID Call Waiting with caller ID Call Hold/Retrieve Three-way conference Calling Line Identity Presentation Call back last missed call Last called number redial User line locking/Call Barring (all current levels) Itemised bill Call Forward Call Forward on No Reply Call Forward on Busy Call Forward Unconditional VoiceMail - Message Waiting Indicator - Call back option Number Portability Secret Number Legal intercept Emergency Number Routing Anonymous Caller Rejection Directory Service update Feature code activation I have looked into both SER and Asterisk, and found out that SER should be the most appropriate platform since most of the users will use a SIP adapter/endpoint and be located on the Internet (like most of the VoIP services these days). So my question to the list is to gather some experiences and possible remarks/comments on which direction I should go :D Any help would be gladly appreciated :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323
Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
Remco Barende wrote: Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Do you have the T in your Dial statment? Remove the T and try it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
FYI, I didn't read your message. With hundreds of messages/day, I use the subject line to decide whether or not to read. Whenever I get a message with (no subject) it is an instant delete. Also, for those of you who think you're still on a 300baud modem and have to conserve every keystroke, whenever I see a u instead of you--instant delete. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
Altus, Yes, Asterisk can do the following scenarios, amongst others: Client -- H.323 -- Asterisk -- H.323 -- Client Client -- H.323 -- Asterisk -- SIP -- Client In these scenarios, it is acting as a Back To Back User Agent (BTBUA). It can also handle video calls, though I have not used this myself. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Altus Snyman wrote: Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clarification on Fax capability?
Wondering if someone (Steve?) can clarify something form me. I think the recent soho fax solution? thread has mixed things up for me. - Is it possible to get reliable fax reception using a Zaptel FXO interface connected to a standard POTS line and a fax machine connected to station interface? - Is it possible to reliably send outboud faxes in the reverse direction? I understand the issues with fax over VoIP. I just want to handle faxes here in my small office without dedicating a line to the machine. The function of faxing over a zaptel interface (eg, x100p, tdm) is not consistent from one implementation to another, regardless of whether your using Stable or Head. The two basic approaches that are commonly discussed on the list are: 1) using Steve's spandsp patches to intercept faxes, creating an *.tif file that can be emailed and viewed outside of *, and, 2) simply switching a fax call through * to a tip/ring interface of some sort that has an attached traditional fax machine. Option #2 works in many cases if the incoming fax call is handled by the g711 codec. The fax call is no different then receiving any other call, however any analog-based modem tones passed through a digital interface (eg, asterisk) are _not_ reproduced reliably. The higher the modem speed, the greater chance of distorting the analog signal tones, and the greater the chance of fax not working at all. (You'll find the older-slower modems will work better then newer-faster fax machines. Essentially, modems that operate at 9600 baud or slower are more reliable then anything faster.) Option #1 works in some cases with Digitum fxo cards, however something close to 50% (or more) implementations fail to work in any form of reliable way. The issuse seems to be related to funcky things happening with the zaptel/wctdm drivers that cause missed pcm frames (or slips) between the digium cards and the asterisk code. Missed or slipped frames will negatively impact any modem-based communications, including fax machines. Steve has received recent reports (#1) that suggest that outgoing faxes are now failing at a significant rate indicating that something has changed in the zaptel/wctdm drivers negatively impacting such calls. No one (to the best of my knowledge) has complained about the drivers impacting voice, however small numbers of missed or slipped frames are generally not noticed (or are not that objectionable). If you dig through the archives, you'll probably notice a fair number of folks having issues with the digium x100p/tdm cards that revolve around interrupt latency and/or pci bus problems. Those issues tend to be associated with echo and many have found that swapping motherboards corrects the problem. But, no one (to the best of my knowledge) has ever discovered why swapping motherboards corrects the problem; they just know that it does. Likewise, no one has assembled a list of motherboards that work simply because it is very difficult to determine motherboard models when companies like Dell HP do not publish who's board they actually use in their products (not to mention that some system vendors have different boards in the exact same system models). We do know that processor speed, number of processors, amount of ram, etc, has little or no impact on the issues. Some have noted that moving from a P4 to a slower speed P3 processor has corrected the issues, but those type changes essentially have hundreds of other changes along with that switch. Its unlikely the actual processor switch had anything to do with it; more likely is that changes that came along in the form of a different pci bus structure actually improved it (or something like that). So, the best guess (today) is that a driver or hardware problem (pci chip set) exists between the digium cards and the underlying motherboard that negatively impacts the reliable transfer of data from those cards to asterisk code, thus impacting voice (echo) and fax (modem signals). It doesn't help that the drivers and cards are essentially considered digium property and that all support should come from that source. Opening a trouble ticket with digium (on this particular issue) tends to go directly into a black hole. Some of that is likely due to the sophistication (and/or lack of understanding) of those responsible for supporting those products since one has to fully understand how to deal with specific hardware (eg, chip sets), kernel drivers, asterisk code, etc. If you try to analyze the code in zaptel and wctdm you'll understand why that is. Bottom line is the tdm card drivers seem to be just okay for voice, but no where near reliable or even predictable for fax. That's based on cvs head and spandsp-pre9 code as of this morning. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
[Asterisk-Users] Integration Panasonic PBX
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
IVR-Voicemail
The Operation:
1)Where a call enters from the PSTN, the PBX flashes and transfer it
to Asterisk.
2)Asterisk receives the call and you head the in the IVR
3)The caller dials the extension number
4)Asterisk will send the call to the extension number dialed before
4.1) if the extension answers, Asterisk should transfer the call and
free the port, leaning the loop formed between the PSTN and the
extension by the PBX and Asterisk ports are left free.
4.2) If the extension doesn't answer or its busy Asterisk will have to
active the voicemail.
For the time being, the inconvenient I've is in the communication with
the PBX, cause Asterisk after sending the sendtdmf loose any contact
with the status of the call.
I need a way to keep control of the extension of the PBX, if it answer
or not or if its busy, so it can passes control to Asterisk, with
another flash command to active the voicemail menu.
This is are example of a dialplan that doesn't works, cause I send the
call to the extension of the PBX, but I don't keep control of the
status of the call but I can't recover it after, cause if I execute
flash again, the control goes back to Asterisk.
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Background(IVR)
exten = s,4,DigitTimeout,4
exten = s,5,ResponseTimeout,4
exten = t,1,Goto(operadora,s,1)
exten = i,1,Playback(invalid)
exten = _1XX,1,Flash
exten = _1XX,2,background(silence/1)
exten = _1XX,3,SendDTMF(${EXTEN})
exten = _1XX,4,background(silence/1)
exten = _1XX,5,Hangup
Thank you
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Re: [Asterisk-Users] Asterisk and Call recognition (call id)
On 10:21, Tue 15 Feb 05, Pablo Fernandes wrote: Hi, Somebody already made call recognition with database access? Depending of call's number, it access a database looking for that number. Where can i find something about this? You can do this with an agi script. It's not that hard to do, depending on your programming skillz. A lot of languages are supported. Have a look at this page: http://www.voip-info.org/wiki-Asterisk+AGI -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm about fed up with SixTel and their vanity number process. I would recommend everyone to stay away from SixTel, for now. I'm just wondering, how long should a vanity number transfer really take? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clarification on Fax capability?
On Tue, February 15, 2005 7:48 am, Rich Adamson said: 2) simply switching a fax call through * to a tip/ring interface of some sort that has an attached traditional fax machine. Does the codec issue with #2 still apply if the incoming fax call is on a Zaptel FXO interface? Is the codec used when connecting two channels on the same zaptel card or does the native bridg[ing] bypass that? Thanks for the info. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension matching in gastman
Sorry for posting before without a subject. Glad to see there are those on the list who do not make mistakes. Diversity keeps things interesting I guess. I have a question for using gastman. I have set up extensions for my IAX users as IAX2/username, and I keep getting the following Dunno how to tell if IAX2/username/6 is IAX2/username I was wondering if there is some sort of wildcard character that can be used here? The number changes every time, so I do not think that I can put in seperate extensions. Thank You, Ron Frederick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
On Tue, February 15, 2005 9:27 am, Rob Risner said: I'm just wondering, how long should a vanity number transfer really take? No help here, just posting a me too to warn others. Friday was 10 days for me. No happy to hear you've waited much longer with the same result. Can never raise them on the phone. They take days to respond to the ticket and are rather terse when they actually do. Not pleased at all. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_rxfax creating bad faxes? (StripOffsets)
Using CVS HEAD (20050214 with the new jitter buffer) and the latest (0.0.6?) spandsp. libtiff version is 3.5.7, compiled from source. System is Slackware 10, 2.4.26 kernel, no fancy patches and processor is a P4 1.5GHz on an Intel motherboard. Most faxes are coming through fine but a few companies in particular are having very poor success faxing to us. Most faxes LOOK like they'd receive correctly, but I get this when trying to convert the TIFF to a pdf: /var/spool/asterisk/faxin/1108478035.15.tiff: TIFF directory is missing required StripOffsets field. What can I do to help debug this? Regards, Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 and/or Euro-ISDN specifications?
Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extra sounds (Weather)
Does anyone know of a AGI script that takes advantage of the weather sound files thats included with the extra sound files available from www.loligo.com/asterisk/sounds/ ? Thank, Jeramie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make of asterisk doesn't do anything...
On Tue, Feb 15, 2005 at 01:16:16PM +, Alistair Cunningham wrote:
Michael,
Someone may know a simple fix. If not, can you please install the
'strace' program, then run:
strace -f -o /tmp/strace.out make
This will run make, and log any system calls it makes to
/tmp/strace.out. When it hangs, take a look in that file. It may have
stopped on one system call, such as select() or poll(), or it may be
scrolling endlessly, repeating the same system calls over and over
again. If grep is using a lot of processor time, it's probably the
latter. Either way, please paste the last 20 or lines into an email, and
post it to this mailing list.
It did just go on and on apparently reading and writing files. It seems to be
complaining alot about unfinished reads and writes... Below are the last
significant 40 lines.
I have saved the output file, so if looking at the first occurrence of PID's
might help, I can look it up.
Thanks!
[pid 22436] ... read resumed
p\5\341\5I\5\367\4\0\6\237\6{\6\t\6`\4\322\2\34\3\256\003..., 24576) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 4096) = 4096
[pid 22440] write(1,
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 4096) = 4096
[pid 22440] read(3,
a\5\376\5\3\6.\7\337\10k\7\7\5I\5j\6*\6~\4\352\2U\3\336..., 4096) = 4096
[pid 22440] write(1,
a\5\376\5\3\6.\7\337\10k\7\7\5I\5j\6*\6~\4\352\2U\3\336..., 4096 unfinished
...
[pid 22436] ... read resumed
\230\6c\2[\377p\376d\377\274\1*\4\204\1:\376\37\373\272..., 16384) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 4096) = 4096
[pid 22440] write(1,
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 4096) = 4096
[pid 22440] read(3,
\226\7\326\3d\0\250\2~\7\6\v\245\f\251\v\224\0105\4\310..., 4096) = 4096
[pid 22440] write(1,
\226\7\326\3d\0\250\2~\7\6\v\245\f\251\v\224\0105\4\310..., 4096 unfinished
...
[pid 22436] ... read resumed
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 8192) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
Q\371u\373\310\371_\364\211\365+\365K\363\355\373\311\3..., 4096) = 4096
[pid 22440] write(1,
Q\371u\373\310\371_\364\211\365+\365K\363\355\373\311\3..., 4096) = 4096
[pid 22440] read(3,
\257\375r\374\340\375+\0j\0\324\377\30\0\242\0\360\0h\1..., 4096) = 4096
[pid 22440] write(1,
\257\375r\374\340\375+\0j\0\324\377\30\0\242\0\360\0h\1..., 4096 unfinished
...
[pid 22436] ... read resumed
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 65536) = 8192
[pid 22436] read(0,
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Mobile operator message
Hello, i using asterisk with DIVA server by CAPI termination. But when i call on off mobile phone, i can listen normaly tone, not operator message about availability user. Can you explain me where are possible mistake? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Rob Risner wrote: I'm just wondering, how long should a vanity number transfer really take? Were you requesting a new vanity number, or a transfer of an existing number? If it's new, have you checked to see if the number is still listed as available? google for vanity toll free number search -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 and/or Euro-ISDN specifications?
On Tue, 15 Feb 2005, Daniel Nyström wrote: Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? The specifications are built one on top of another. Each just lists the changes and clarifications relative to the underlaying specification. E.g. for the Swedish incumbent operator Telia you have: * Telia ISDN Notes for Suppliers * ETSI specifications (e.g. ETSI ETS 300 403 01 etc) * ITU specifications (e.g. Q.931, Q.921 and lots and lots of others) The Telia implementation specification is free, as are the ETSI specification (at least for some uses). The ITU specifications are a bit expensive, but you can download three for free. Most of the time the ITU specifications are at the bottom level, but not always. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, inband DTMF send by a GSM mobile
Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Same boat here. Actually got someone on AOL instant messenger yesterday. Their response as follows when asked how long it will take to get our 800 number: [15:11] sixtel9: it's in the works any time frame? [15:14] sixtel9: not specifically, we switched carriers so we're dealing w/ some issues just need to know if it will be weeks/months/ or days [15:21] sixtel9: days - Pedro On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas [EMAIL PROTECTED] wrote: On Tue, February 15, 2005 9:27 am, Rob Risner said: I'm just wondering, how long should a vanity number transfer really take? No help here, just posting a me too to warn others. Friday was 10 days for me. No happy to hear you've waited much longer with the same result. Can never raise them on the phone. They take days to respond to the ticket and are rather terse when they actually do. Not pleased at all. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clarification on Fax capability?
On Tue, February 15, 2005 7:48 am, Rich Adamson said: 2) simply switching a fax call through * to a tip/ring interface of some sort that has an attached traditional fax machine. Does the codec issue with #2 still apply if the incoming fax call is on a Zaptel FXO interface? Is the codec used when connecting two channels on the same zaptel card or does the native bridg[ing] bypass that? Funny that you should ask. I just finished testing it using the faxdetect=incoming method, redirecting the call to a Cisco ata186. The incoming call arrived on a TDM fxo port. Received the fax header, but the remainder of the page was blank. The sender received a message indicating a failure. Best guess... the tdm driver problem is impacting the ability to send the fax tones reliably even with g711. Its very likely to be the interrupt latency and/or pci bus problem on this particular system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 question
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge.
I patched the code due so that Lucent can handle asterisk's ver4 h323
http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration
I can now successfully dial in to our company over multiple lines.
The issue is when I dial out
The first outgoing call connects to an outside user A
The second call drops the first user and connects the second call to the
outside user A
I'm not sure if it's a problem with my configuration or if the gatekeeper
isn't handling the calls correctly.
Executing Dial(SIP/108-4ec4, OH323/3758112) in new stack
-- H.323 call to 3758112 with codec(s) ulaw
-- Called 3758112
-- OH323/L15517 answered SIP/108-4ec4
-- H.323 call 'ip$localhost/15516' cleared, reason 8 (Transport failure)
-- Hungup 'OH323/L15516'
The call is dropped with a reason 8 (Transport failure)
the relevant portion of the extensions.conf file is
[macro-dialout]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) ;check
for CID override
for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6)
;check for CID override for
trunk
exten = s,5,SetCallerID(${OUTCID_${ARG1}})
exten = s,6,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten = s,7,Dial(OH323/${ARG2:${length}})
exten = s,8,Congestion
exten = s,108,Macro(outisbusy)
the relevant portion of the extensions.conf file is
[macro-dialout]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) ;check
for CID override
for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6)
;check for CID override for
trunk
exten = s,5,SetCallerID(${OUTCID_${ARG1}})
exten = s,6,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten = s,7,Dial(OH323/${ARG2:${length}})
exten = s,8,Congestion
exten = s,108,Macro(outisbusy)
; dialout using default OUT trunk - no prefix
[macro-dialout-default]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) ;check
for CID override
for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID} = foo]?6) ;check for CID
override for trunk
exten = s,5,SetCallerID(${OUTCID})
exten = s,6,Dial(OH323/${ARG1})
exten = s,7,Congestion
exten = s,107,Macro(outisbusy)
and oh323.conf is
general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=no
h245inSetup=no
inBandDTMF=yes
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=2
libTraceLevel=2
libTraceFile=stdout
; ip address changed to protect the innocent
gatekeeper=66.173.aaa.bbb
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
context=from-pstn
[register]
alias=757747
alias=7575881112
alias=757533
alias=7575831114
alias=757535
[codecs]
codec=G711U
frames=20
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Re: [Asterisk-Users] Sip phones how to dial a # sign?
Is there a way to somehow do an escape # so that you can still use the # key to control devices that require a #, but still keep the T in the dial plan? We have clients that need to check external voicemail systems that require the use of the # sign, but still want to have the call parking feature. On Tue, 15 Feb 2005 06:54:23 -0700, Michael Welter [EMAIL PROTECTED] wrote: Remco Barende wrote: Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Do you have the T in your Dial statment? Remove the T and try it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_rxfax creating bad faxes? (StripOffsets)
Using CVS HEAD (20050214 with the new jitter buffer) and the latest (0.0.6?) spandsp. libtiff version is 3.5.7, compiled from source. System is Slackware 10, 2.4.26 kernel, no fancy patches and processor is a P4 1.5GHz on an Intel motherboard. Most faxes are coming through fine but a few companies in particular are having very poor success faxing to us. Most faxes LOOK like they'd receive correctly, but I get this when trying to convert the TIFF to a pdf: /var/spool/asterisk/faxin/1108478035.15.tiff: TIFF directory is missing required StripOffsets field. What can I do to help debug this? Over what facility are the incoming faxes arriving (pri, isdn, tdm)? BTW, spandsp-pre9 is the most current. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA that actually work with T.38
On Feb 14, 2005, at 5:27 PM, Cory Andrews wrote: There is just a form that needs to be completed, which we forward on to Linksys and they approve or deny the application based upon the background of the applicant. Have had very few applications rejected, pretty straightforward process. I don't get it. My assumption is that by background you mean the applicant turns out to be a VOIP provider as opposed to say a lot of people on this list that would be interested in buying one of these devices but can't because they're really just end users. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas [EMAIL PROTECTED] wrote: On Tue, February 15, 2005 9:27 am, Rob Risner said: I'm just wondering, how long should a vanity number transfer really take? No help here, just posting a me too to warn others. Friday was 10 days for me. No happy to hear you've waited much longer with the same result. Can never raise them on the phone. They take days to respond to the ticket and are rather terse when they actually do. Not pleased at all. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, inband DTMF send by a GSM mobile
On Feb 15, 2005, at 10:09 AM, Florian Lefeuvre wrote: Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this I suffer from this problem all of the time but I'm configured for out-of-band dtmf on the asterisk side thanks to IAX2 trunking. I think the problem is GSM or maybe cell phones in general. I know this has been discussed before. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA that actually work with T.38
Mark Eissler wrote: On Feb 14, 2005, at 5:27 PM, Cory Andrews wrote: There is just a form that needs to be completed, which we forward on to Linksys and they approve or deny the application based upon the background of the applicant. Have had very few applications rejected, pretty straightforward process. I don't get it. My assumption is that by background you mean the applicant turns out to be a VOIP provider as opposed to say a lot of people on this list that would be interested in buying one of these devices but can't because they're really just end users. Nah. The people who can't get approval are reprobates like, drunks, hobos and an CEOs of large corporations. They are just eager to ensure a better class of customer. :-) Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. Well let me tell you one thing about that, whenever a VOIP provider runs out of DIDs in their pool, it can take days. I had the same problem with Voicepulse several weeks ago where I ordered two DIDs in the same exchange. Got the first number immediately but the second one took somewhere between 4 to 6 weeks. Come to think of it, the exact same problem happened to me with Broadvox almost exactly one year earlier. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extra sounds (Weather)
How about writing some script that works with a free service like weather.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeramie Rentfrow Sent: Tuesday, February 15, 2005 9:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Extra sounds (Weather) Does anyone know of a AGI script that takes advantage of the weather sound files thats included with the extra sound files available from www.loligo.com/asterisk/sounds/ ? Thank, Jeramie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Users in Madrid?
I'm sitting in a hotel close to the Madrid airport... Any Asterisk users in the neighbourhood that wants to meet me for a beer and some Asterisk hacking this evening? Send e-mail to me *off list*, thank you. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't run AGI for outbound call
Thanks Stefan - works like charm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter Sent: 15. febrúar 2005 00:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't run AGI for outbound call On Tue, 2005-02-15 at 00:07 +, Ívar Ragnarsson wrote: The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my NetMeeting client Channel: h323/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Application: AGI(agi-test.agi) Data: 1234 [SNIP] Feb 14 23:53:25 WARNING[7958]: pbx.c:4164 ast_pbx_run_app: No such application 'AGI(agi-test.agi)' Asterisk is looking for an application called 'AGI(agi-test.agi)' and that one obviously does not exist. The Application property must only contain the name of the application, i.e. AGI. Any parameters are given via the Data property. What you probably want is: Channel: h323/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Application: AGI Data: agi-test.agi stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed
Peer Oliver Schmidt posde-at-theinternet.de |Asterisk/Maestro| wrote: A. Peverelli wrote: I own a ME600 EPIA Mini-ITX main board with the latest Debian distro (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, isdnactivecards installed. I have a QuadBRI module by Junghanns with bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, but I have some strange behaviour. All modules seems to be correctly installed and actives, but on /dev I find only capi20. Anyway, starting Asterisk, I recevive a 'CAPI not installed!' error on chan_capi load and I can't find why. Anyone has some idea? quadBRI CAPI!!! The quadbri cards do not use/support CAPI. If you don't have another CAPI capable device in your system you can't/shouldn't use CAPI (I guess you could use CAPI via mISDN, but what is the point?) Thank-you very much! Your answer made me understand many things! So... the point is that I have a Linux application CAPI speaking and I would like to connect it with Asterisk. Another goal is to connect Asterisk with an ISDN PBX, so I thought that I may do both things at once. Now I think that I have to change some architectural parameter... If someone has any suggestion on that, I will really appreciate it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autostart Asterisk on Slackware?
Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autostart Asterisk on Slackware?
Goran Dj. wrote: Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? In my /etc/rc.d/rc.local # Put any local setup commands in here: /sbin/ztcfg /etc/rc.d/rc.hdlc /usr/sbin/asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home .5 Setup help with 4 X100P
Hello All, I installed [EMAIL PROTECTED] .5 last night. I was able to configure some extensions for the house and they work fine. I just can't make inbound and/or outbound calls. The Flash Operator Panel shows four external icons and my new extensions. I have four X100P and two Broadvoice sip accounts. Thanks, David -- David Shaw [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autostart Asterisk on Slackware?
On February 15, 2005 10:49 am, Goran Dj. wrote: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? Something like /usr/sbin/asterisk -g in /etc/rc.d/rc.local would do it. You can craft up more complex things if you like, wrap safe_asterisk or do whatver, but that'll get you started. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clarification on Fax capability?
I've been trying this for a while and I have been unable to get a reliable connection betwen two Zaptel FXS interfaces, so the bridging does afect data transfer. Anybody got some tunning tips to get this to work ? I'm using a dual PIII with a ServerWorks Chipset, two TDM cards (8xFXS) and a Fritz Capi to connect to my telecom provider. I can send faxes with some success, but receiving rate of success is less than 30%. Fax information for Asterisk is difficult to come by is everybody using spandsp's way ? Thx Pedro Caria On 15/fev/2005, at 15:05, Rich Adamson wrote: On Tue, February 15, 2005 7:48 am, Rich Adamson said: 2) simply switching a fax call through * to a tip/ring interface of some sort that has an attached traditional fax machine. Does the codec issue with #2 still apply if the incoming fax call is on a Zaptel FXO interface? Is the codec used when connecting two channels on the same zaptel card or does the native bridg[ing] bypass that? Funny that you should ask. I just finished testing it using the faxdetect=incoming method, redirecting the call to a Cisco ata186. The incoming call arrived on a TDM fxo port. Received the fax header, but the remainder of the page was blank. The sender received a message indicating a failure. Best guess... the tdm driver problem is impacting the ability to send the fax tones reliably even with g711. Its very likely to be the interrupt latency and/or pci bus problem on this particular system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link
On Tue, 15 Feb 2005, Sylvain Gagnon wrote: I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the monitor command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. I'm not quite sure I understand your setup when the dtmf detection works / does not work. Can you explain what is connected to what for the two cases? One possibility I can think of is that inband digits are not detected on a PRI until the call is in the PROCEEDING phase. Until then the digits are dent as INFORMATION elements. Perhaps your two pieces of equipment do not agreee on the state of the call? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting a Forward to an external number on yourphone
[EMAIL PROTECTED] wrote:
Hi!
Maybe I have just been looking on the wrong pages but there is a
question that is very important for me. I already studied some
Demo-Dialplans and made some basic experiences with Asterisk.
But what I
need to find out is how I can handle this.
I am leaving my office and I want to tell asterisk to forward
calls now
to my mobile phone by just hitting a key (on my IP-Phone) or
by using a
special key-sequence.
How can this be handled because I need to change the dialplan
based on
some information coming from a device attached to a channel.
When back in my office I hit the key again and the calls are
now routed
to my IP-Phone (or ISDN-Phone on zap-channel) again.
With IP-Phones I can imagine just unregistering the phone and
having a
dialplan with a fallback-option or something like that. But what if I
want to tell asterisk to forward calls from now on to a
number I want to
manually add just for today (hitting a key, entering the new target
number and that's it). where can I find some information on
how to make
this feature available.
There's probably a whole lot of ways this could be achieved. It's kind
of a cookbook type thing: more than one recipe.
What I've been working on is a way to change where zero-out or main
menu timeouts. I want to be able to do this while on the road or in the
office, so I built it into the dial plan.
All the authentication issues aside, I need to set a variable that
defines where I want calls to go. Additionally, I want the state of this
variable to survive a restart.
What I've been messing with is using a file-based semaphores to do this.
Perhaps it's kludgy, but it works. Also, there's no database work
required: it all happens in the dialplan.
There's a lot more work needed here - I just hacked this together. Works
pretty well though, it's just not very friendly.
I wouldn't intall this at a customer without some more work, but mostly
because I'd want it a bit more friendly.
Anyhow, I have not tested this exact one, but it's similar to what I've
got so far. Enjoy:
[global]
MyCell=18005551212
Reception = SIP/YourPhone
; These are your semaphores. Because they are files they will survive a
restart
#include /var/lib/asterisk/remote_context
#include /var/lib/asterisk/remote_exten
[incoming]
; somewhere you'll need to put the exten that sends you to [presence]
; Make sure this is secure, but you'll probably want to be able to
; access it extenally
exten = 4321,1,Goto(presence,s,1)
[local_sets]
exten = 1000,1,Macro(exec_set,${EXTEN},${Reception})
[remote_sets]
exten = 2000,1,Macro(cell_user,${MyCell})
[presence]
;just record a basic prompt so you know what's up
exten = s,1,SayDigits(${ATDT_EXTEN}) ; raw, but it'll tell you where
it's going
exten = s,2,Background(prompt_for_choice)
this sets calls to go to one place
; first we write the semaphore
exten = 1,1,System(echo ATDT_CONTEXT=remote_sets
/var/lib/remote_context)
;then we set the value of the variable
exten = 1,2,SetGlobalVar(ATDT_CONTEXT=remote_sets)
exten = 1,3,System(echo ATDT_EXTEN=2000 /var/lib/remote_exten)
exten = 1,4,SetGlobalVar(ATDT_EXTEN=2000)
exten = 1,5,Goto(presence,s,1)
this sets calls to go to somewhere else
exten = 2,1,System(echo ATDT_CONTEXT=local_sets
/var/lib/remote_context)
exten = 2,2,SetGlobalVar(ATDT_CONTEXT=local_sets)
exten = 2,3,System(echo ATDT_EXTEN=1000 /var/lib/remote_exten)
exten = 2,4,SetGlobalVar(ATDT_EXTEN=1000)
exten = 2,5,Goto(presence,s,1)
you can have as many of these as you need
exten = 3,1,System(echo ATDT_CONTEXT=[some_context]
/var/lib/remote_context)
exten = 3,2,SetGlobalVar(ATDT_CONTEXT=[some_context])
exten = 3,3,System(echo ATDT_EXTEN=[some_exten]
/var/lib/remote_exten)
exten = 3,4,SetGlobalVar(ATDT_EXTEN=[some_exten])
exten = 3,5,Goto(presence,s,1)
[cleanup]
; hang up the call and whatever else
exten = s,1,Playback(goodbye)
exten = s,2,Hangup()
;;; MACROS ;;;
[macro-exec_set]
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20
seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as
no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send
the user into VoicemailMain
[macro-cell_user]
; This will only work if your external line supports link transfer
exten = s,1,Playback(transfer)
exten = s,2,Flash()
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
--
No virus found in this outgoing message.
Checked by AVG
Re: [Asterisk-Users] Dlink VPNs??
On Sun, 13 Feb 2005 13:39:09 -0500, Mike Chapman [EMAIL PROTECTED] wrote: Hi, I am thinking of purchasing a cheap Dlink VPN for testing purposes for use with my Asterisk box and would like to ask the list for advice on how to pick a VPN that will work with my box. I am a newbie to both VPN's and Asterisk so any advice will be appreciated. Thanks, Mike ___ I have 2 clients using the 3com secure gateways... never had a problem yet! My cisco 2610 and pix hate it... but some people have luck with them. -- A.G. (Tony) Nichols I.S. Manager ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts??
http://advancedippipeline.com/60400413 BOULDER, Colo. -- Leading Voice over IP service provider Vonage Holdings has complained to the Federal Communications Commission that competitors are blocking the use of its service, according to FCC chairman Michael Powell and others close to the company. We're very actively on this case and we are taking it pretty seriously, said Powell, during an interview Monday here at the Silicon Flatirons conference. In a speech at the conference Sunday, Stanford law professor Larry Lessing said that Vonage has been telling the FCC that other service providers are hampering Vonage's VoIP service by blocking it from reaching certain SIP addresses for end-user devices ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Sam thing here. Waiting 10+ business days for my DID. Can't get through to them by phone and email responses take days. These guys are worthless. On Tue, 15 Feb 2005 10:40:44 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri lin k
Title: RE: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link Thank you Peter for you reply, I realize this problem occur because I take the CVS head (maybe a bugs get introduce), because when I rebuild using the checkout of the latest stable version (cvs checkout -r v1-0), I don't have the problem of detection of DTMF over a direct ISDN pri link. Hopefully this problem with be fixed before the next release of asterisk. Sylvain. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Peter Svensson Sent: Tuesday, February 15, 2005 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link On Tue, 15 Feb 2005, Sylvain Gagnon wrote: I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the monitor command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. I'm not quite sure I understand your setup when the dtmf detection works / does not work. Can you explain what is connected to what for the two cases? One possibility I can think of is that inband digits are not detected on a PRI until the call is in the PROCEEDING phase. Until then the digits are dent as INFORMATION elements. Perhaps your two pieces of equipment do not agreee on the state of the call? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
On Tue, 2005-02-15 at 13:59 +, Alistair Cunningham wrote: It can also handle video calls, though I have not used this myself. AFAIK video only with SIP, which I didn't test myself either. With H323 it does not work, audio only there. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7912G via SIP, looking for comments
Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems?Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation like this (built in switch, display, speaker phone...) Thanks Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 206.666.1786 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Integration Panasonic PBX
Maximiliano,
We have implemented that solution succesfully several times.
First:
Does your Panasonic support dtmf inband signaling? without that forget it.
Also you need your setup to look like this:
Outside calls ring into pbx. Pbx co lines are forwarded to a group of
extensions set as voice mail extensions in the pbx programming. Those
extensions are connected to asterisk via an fxo card. That way
asterisk can do ivr and voicemail. You also need to program the pbx
having all phones set to forward all calls (when nobody picks up or is
busy) to that group of extensions so that the call comes back to *
carrying the id of the extension with it.
That's it.
If you need some help let us know. We are in Argentina.
--
Sergio Veltri
www.pointhorizon.com
Tel: +5411-5217-1295
Cell: +54-911-5604-4149
Message: 7
Date: Tue, 15 Feb 2005 11:09:04 -0300
From: Maximiliano J. Goldsmid [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration Panasonic PBX
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=UTF-8
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
IVR-Voicemail
The Operation:
1)Where a call enters from the PSTN, the PBX flashes and transfer it
to Asterisk.
2)Asterisk receives the call and you head the in the IVR
3)The caller dials the extension number
4)Asterisk will send the call to the extension number dialed before
4.1) if the extension answers, Asterisk should transfer the call and
free the port, leaning the loop formed between the PSTN and the
extension by the PBX and Asterisk ports are left free.
4.2) If the extension doesn't answer or its busy Asterisk will have to
active the voicemail.
For the time being, the inconvenient I've is in the communication with
the PBX, cause Asterisk after sending the sendtdmf loose any contact
with the status of the call.
I need a way to keep control of the extension of the PBX, if it answer
or not or if its busy, so it can passes control to Asterisk, with
another flash command to active the voicemail menu.
This is are example of a dialplan that doesn't works, cause I send the
call to the extension of the PBX, but I don't keep control of the
status of the call but I can't recover it after, cause if I execute
flash again, the control goes back to Asterisk.
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Background(IVR)
exten = s,4,DigitTimeout,4
exten = s,5,ResponseTimeout,4
exten = t,1,Goto(operadora,s,1)
exten = i,1,Playback(invalid)
exten = _1XX,1,Flash
exten = _1XX,2,background(silence/1)
exten = _1XX,3,SendDTMF(${EXTEN})
exten = _1XX,4,background(silence/1)
exten = _1XX,5,Hangup
Thank you
--
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Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts??
You can always visit Slashdot for countless (useless, well, not always) comments: http://yro.slashdot.org/article.pl?sid=05/02/14/2352254 --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP Serer ????
Is the Cisco phone book via XML something specific to [EMAIL PROTECTED], or is this something that can be implemented within a normal Asterisk deployment..? Thanks On Mon, 14 Feb 2005 17:43:36 -0800 (PST) [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] has Asterisk, a TFTP server, and a web based cisco phone config tool. It auto installs it all. This should save you a lot of time. you can be up and running in an hour. It also has a built in phone book cisco XML service that works well with the 7960. http://asteriskathome.sourceforge.net/ --- Stefan Gofferje [EMAIL PROTECTED] wrote: Ferguson, Michael schrieb: G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Any Linux distro should ship with one or two tftp servers. Anyway, away from firmware updates, the config could be done via phone menu or webinterface. There also are various tftpds available for Windows. BTY: Does anyone have a How-To on getting the 7960 fully configured for *? http://www.voip-info.org/tiki-index.php?page=cisco%2079xx http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960 Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. Did you get any notice from them on the DID? The dropdown for unlimited use DIDs only gives a choice for Area Code. Have you had any communication with them on the actual prefix you wanted? After clicking the button myself, I eventually found out that they couldn't give me a DID that was local to me. I had one other ticket open that now seems to be MIA. I'll not be using them for anything real important. There doesn't seem to be a business in the market that one could rely on. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite Softphone
Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Comments on Vonage SIP port blocking com plai nts??
Yeah, I'd like to hear you guys' opinion instead of CleverNickName's! -Original Message- From: Luki [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 15, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts?? You can always visit Slashdot for countless (useless, well, not always) comments: http://yro.slashdot.org/article.pl?sid=05/02/14/2352254 --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting SPEEX to work
-Original Message- Sorry to bump this post, but I'm losing my mind a bit on why Speex won't negotiate for me. Is _anyone_ using it out there? Here's my original query with the CLI log http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088225.html I hate to check the obvious, but is speex allowed in your channel configuration ? (allow=speex) Absolutely. Don't ever worry about not stating the obvious! That's what fora are for... to watch each others' backs. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
I've used them too and got absolutely nothing from them. My e-mails hardly ever get responses and when they do respond, it's usually a one-liner that evades the question. Stay as far away as you can from Sixtel / IAX.cc. I think a BBB complaint about them should be made. Mohit. On Tue, 15 Feb 2005 11:52:33 -0500, Andrew Thompson [EMAIL PROTECTED] wrote: BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. Did you get any notice from them on the DID? The dropdown for unlimited use DIDs only gives a choice for Area Code. Have you had any communication with them on the actual prefix you wanted? After clicking the button myself, I eventually found out that they couldn't give me a DID that was local to me. I had one other ticket open that now seems to be MIA. I'll not be using them for anything real important. There doesn't seem to be a business in the market that one could rely on. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Integration Panasonic PBX
will be nice to have this setting posted. Here in Panama we use lots
of Panasonics and that is a nice one to have
Cualquier cosa nueva me la hacen saber por este posting o a mi correo
eaperezh @ gmail.com
Saludos,
On Tue, 15 Feb 2005 13:38:26 -0300, Sergio Veltri
[EMAIL PROTECTED] wrote:
Maximiliano,
We have implemented that solution succesfully several times.
First:
Does your Panasonic support dtmf inband signaling? without that forget it.
Also you need your setup to look like this:
Outside calls ring into pbx. Pbx co lines are forwarded to a group of
extensions set as voice mail extensions in the pbx programming. Those
extensions are connected to asterisk via an fxo card. That way
asterisk can do ivr and voicemail. You also need to program the pbx
having all phones set to forward all calls (when nobody picks up or is
busy) to that group of extensions so that the call comes back to *
carrying the id of the extension with it.
That's it.
If you need some help let us know. We are in Argentina.
--
Sergio Veltri
www.pointhorizon.com
Tel: +5411-5217-1295
Cell: +54-911-5604-4149
Message: 7
Date: Tue, 15 Feb 2005 11:09:04 -0300
From: Maximiliano J. Goldsmid [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration Panasonic PBX
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=UTF-8
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
IVR-Voicemail
The Operation:
1)Where a call enters from the PSTN, the PBX flashes and transfer it
to Asterisk.
2)Asterisk receives the call and you head the in the IVR
3)The caller dials the extension number
4)Asterisk will send the call to the extension number dialed before
4.1) if the extension answers, Asterisk should transfer the call and
free the port, leaning the loop formed between the PSTN and the
extension by the PBX and Asterisk ports are left free.
4.2) If the extension doesn't answer or its busy Asterisk will have to
active the voicemail.
For the time being, the inconvenient I've is in the communication with
the PBX, cause Asterisk after sending the sendtdmf loose any contact
with the status of the call.
I need a way to keep control of the extension of the PBX, if it answer
or not or if its busy, so it can passes control to Asterisk, with
another flash command to active the voicemail menu.
This is are example of a dialplan that doesn't works, cause I send the
call to the extension of the PBX, but I don't keep control of the
status of the call but I can't recover it after, cause if I execute
flash again, the control goes back to Asterisk.
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Background(IVR)
exten = s,4,DigitTimeout,4
exten = s,5,ResponseTimeout,4
exten = t,1,Goto(operadora,s,1)
exten = i,1,Playback(invalid)
exten = _1XX,1,Flash
exten = _1XX,2,background(silence/1)
exten = _1XX,3,SendDTMF(${EXTEN})
exten = _1XX,4,background(silence/1)
exten = _1XX,5,Hangup
Thank you
--
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--
---
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
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[Asterisk-Users] IAX2 bugs...
Has anyone had stability issues with IAX2. (Asterisk 1.0.5). reddwarf*CLI iax2 show firmware Device Version Size iaxy 22 39344 I'm asking because in the last three weeks I've noticed the following two issues (on separate occasions): 1) Placed a phone call. Asterisk logs show the phone being answered and various files being Played back. But can't hear anything over the phone. 2) Placed a phone call. Pause. Busy tone. Asterisk never gets the call. iax2 show registry shows the connection (with the service provider) as Registered. Both times, restarting Asterisk has solved the problem. Of course, I'm not happy with this solution as I'm trying to provide a 24hr service here. Could this be a service provider problem? Mohit. -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite Softphone
Hi while on a call.. did you check your CPU usage.. i have a P3 and sometimes when i move my mouse, xlite starts to stutter.. cpu then running 100% just my 2cents chow L - Original Message - From: Richard J. Sears [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 15, 2005 6:56 PM Subject: [Asterisk-Users] X-Lite Softphone Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
