Re: [asterisk-users] Anyone have a reliable T.38 Solution
On 04.01.2012 07:25, Matt Darnell wrote: We are looking to roll a solution that will have the following network layout: ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! we are using such a setup: Telco -PRI- bero*fixBox -BRI- Fritz!BoxFon -analog- FaxMachine | +---SIP- Asterisk -SIP- Phones | +--T.38- t38modem -tty- Hylafax The bero*fixBox http://www.beronet.com/product/berofix-gateways/ is has one ISDN-PRI and 4 ISDN-BRI ports. The PRI-port is connected to the telco, one of the ISDN-BRI ports is connected to an old Fritz!Box Fon http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon/index.php which has 3 (newer ones only 2) analog ports. The calls are routed by the bero*fixBox to either the asterisk server or the hylafax server based on the DID number. The bero*fix box has ~30 softmodems that translate to T.38-SIP which is then connected to the t38modem on the hylafax server which handles around 70,000 faxpages a month. Outgoing (incoming only for test purposes) faxes from our analog fax machines are routed by our asterisk server back to the bero*fix box so we achive bit-slip-free (and therefore reliable) connection. Beronet also has PCI- and PCIe-cards to be directly installed in your asterisk server. Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge 10 How can I playback a soundfile to an existing conference
Hi,
i'm trying to periodically playback a sound to an existing conference
with ConfBridge on Asterisk 10.0.0-rc3
Previously with MeetMe I generated a callout file and had an matching
local dialplan entry.
But this does not work... The local channel gets joined to the
conference, is stuck there until kicked and no sound is played back.
CallerID: TEST 08154711
Channel: Local/123@patience/n
Context: patience
Extension: s
MaxRetries: 0
Priority: 1
WaitTime: 5
[patience]
exten = s,1,Answer()
same = n,Playback(please-be-patient)
same = n,Hangup()
exten = _X.,1,ConfBridge(${EXTEN})
same = n,Hangup()
--
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[asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing
hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips. as long as the ports are on the same pci-card, they're synchronous, but not when one has to use another card (e.g. having a PRI telco line and some analog fax machines or some BRI ISDN equipment served by asterisk) junghanns or beronet have a solution for this (PCM port on the card; they can even switch the voicedata over this bus), but i can't find any solution for digium cards. i've found a timing connector on the TE410P. can this somehow be utilized? is there another (software) solution? am i the only one with this problem (haven't found anything about this on the mailing list) thanks frank sautter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.1 + TDM400P + fax machine unreliable ?
Alex Ongena wrote: I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400Ptab=support (see the last list item) in other words: it often does not work. regards frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: AoC (Advice of Charge)
hello tomislav, Tomislav Parcina wrote: Does Asterisk support Advice of Charge? I was told that my Telco sends me billing signalization that way, and I wonder can I use it? I have found out that this is part of EURO ISDN standard. q.956 - Advice Of Charge. Does anybody know how to implement this with Asterisk? I would like to store those informations (that I recive from my telco by q.956 standard) in MySQL, csv or any other format. i have _partially_ implemented AOC into the libpri and chan_zap part of asterisk (the IEs for AOC units are decoded and encoded and you will see the AOC info on the console if you have increased verbosity to 5). unfortunately it was beyond my scope to propagate this information to the bridged channel, as the info from the telco provider is transmitted during the call termination phase and asterisk destroys the bridge to early (right after it receives the first notice that the call has to be terminated) and so there is no possibility to pass the AOC to the bridged channel nor to write anything to the CDR. what has to be done is to rewrite the call termination process so it does not terminate each of it's bridged legs seperately in a state machine but together. regards frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup with Dialog on snom display
hello bastian, you could use the patch i made http://bugs.digium.com/view.php?id=5014 frank Bastian Schern schrieb: I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via hint priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: SNOM360 Monitoring Extension States
Jason Pyeron schrieb: On Wed, 9 Nov 2005, Olle E. Johansson wrote: That is not supported yet. There is a patch in the issue tracker that does this, but it's a proof-of-concept code. It will burden your asterisk quite a lot if you put it to use in larger production sites. Which issue are you refering to? i think he refers to my patch: http://bugs.digium.com/view.php?id=5014 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***
Olle E. Johansson wrote: We really need test input of the latest patch in this issue report. And we need them today. If you are interested in device state notification in SIP - stop whatever you are doing and give us feedback NOW! Thank you for your assistance! http://bugs.digium.com/view.php?id=3644 PS. Thanks to Xylome for updating this patch so many times! i can only second olle! this patch has a track record since february and i know from all the emails i received from users and the posting to the digium lists that this is an essential feature! so if you need this functionality give feedback to the bugtracker and it will be gladly in asterisk 1.2. regards frank (aka xylome) ___ --Bandwidth and Colocation sponsored by Easynews.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] Snom 360 - Message waiting and conference keys
Paul Brock schrieb: Trying to set up these two buttons on a snom 360. The message waiting key seems to send a call to it's own number, which is obviously engaged and where you are prompted to leave another message to yourself, and the conference key seems to do nothing. this should no longer be a problem, as my patch http://bugs.digium.com/view.php?id=4801 found it's way into cvs... use the latest cvs-head and the new sip.conf option vmexten (this sets the Message-Account in the MWI notify. regards frank ___ 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] compiling CVS-HEAD + Patch from http://bugs.digium.com/view.php?id=3644
Christian Wengel schrieb: But now I have another problem. The LEDs on the snoms are blinking now, if the extension is ringing. But I can't pickup the call by hitting the blinking button. this problem is not solved by only applying patch #3644. but as you are not the first one asking for this... try this one http://bugs.digium.com/view.php?id=5014. from now on i will submit my experimental features frequently to this bug-id. regards frank ___ 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] compiling CVS-HEAD + Patch from http://bugs.digium.com/view.php?id=3644
hi christian, Christian Wengel schrieb: But the latest patch sipsubscribe-20050812.rev806v2.txt from http://bugs.digium.com/view.php?id=3644 didn't worked, maybe you like to try the latest patch i created a view hours ago... sipsubscribe-20050823.rev813.txt on http://bugs.digium.com/view.php?id=3644 regards frank ___ 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] TE405P / TE410P with 2nd generation firmware field upgradable?
hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime odbc/mysql eating connections
Matthew Boehm wrote: Since you are using ODBC, this seems more likely to be an ODBC issue. If you are concerned, you should just use the native MySQL RealTime driver. It does not exibit the behavior you mentioned. Frank Sautter wrote: our asterisk is configured to retrieve sippeers and iaxpeers via odbc from a mysql database. after each call show processlist; within the mysql console shows 2 more persistent connections which are showing no further activity and will not go away even after restaring asterisk. well after changing from res_odbc to res_mysql and cdr_odbc to cdr_mysql this problem was gone. but after i looked if everything was working ok, i found my real problem: the cdr database was somehow corrupted and i had to make a 'myisamchk --recover'! regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime odbc/mysql eating connections
our asterisk is configured to retrieve sippeers and iaxpeers via odbc from a mysql database. after each call show processlist; within the mysql console shows 2 more persistent connections which are showing no further activity and will not go away even after restaring asterisk. is anybody else experiencing this? what can i do do resolve this? this is a show processlist on the mysql console +-+--+---+--+-+---+---++ | Id | User | Host | db | Command | Time | State | Info +-+--+---+--+-+---+---++ | 7 | asterisk | localhost | asterisk | Sleep | 2 | | NULL | 8 | asterisk | localhost | asterisk | Sleep | 13596 | | NULL| 11 | asterisk | localhost | asterisk | Sleep | 13596 | | NULL . stuff deleted ... | 171 | asterisk | localhost | asterisk | Sleep | 31| | NULL | 172 | asterisk | localhost | asterisk | Sleep | 31| | NULL | 173 | asterisk | localhost | asterisk | Sleep | 1 | | NULL | 174 | asterisk | localhost | asterisk | Sleep | 1 | | NULL +-+--+---+--+-+---+---++ 160 rows in set (0.00 sec) # less /etc/odbc.ini [asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User= asterisk Password= obscured #Port = 3306 Database= asterisk # less /etc/asterisk/res_odbc.conf [asterisk] dsn = asterisk username = asterisk password = obscured pre-connect = yes # less /etc/asterisk/extconfig.conf [settings] iaxusers = odbc,asterisk,iaxfriends iaxpeers = odbc,asterisk,iaxfriends sipusers = odbc,asterisk,sipfriends sippeers = odbc,asterisk,sipfriends ___ 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] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
Maik Schmitt schrieb: one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco ---pri--- asterisk ---pri--- legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with 'overlapdial=yes'? This is normal behaviour if you use '.' in your extensions.conf. Use '!' instead and Asterisk will start dialing immediately. when i change '.' to '!' then the overlap digits get lost. this means the longest number dialled on my telco line is as long as there are abigous matches in the dialplan. isn't there a way to start dialling after one received enough digits to decide which path to dial and then still transmit the remaining (overlapping) digits? regards frank ___ 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] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco ---pri--- asterisk ---pri--- legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with 'overlapdial=yes'? here is an excerpt from the logfile (i assume the number is dialled enbloc as it come with the redial function of the legacy pbx): 2005-07-28 17:23:37 VERBOSE[13873] logger.c: -- Accepting overlap call from '070314161XXX' to 'unspecified' on channel 0/1, span 2 2005-07-28 17:23:37 VERBOSE[13997] logger.c: -- Starting simple switch on 'Zap/32-1' 2005-07-28 17:23:40 VERBOSE[13997] logger.c: -- Executing Dial(Zap/32-1, ZAP/g1/0172XXX) in new stack 2005-07-28 17:23:40 VERBOSE[13997] logger.c: -- Requested transfer capability: 0x00 - SPEECH 2005-07-28 17:23:40 VERBOSE[13997] logger.c: -- Called g1/0172XXX 2005-07-28 17:23:41 DEBUG[13872] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 2005-07-28 17:23:41 VERBOSE[13997] logger.c: -- Zap/1-1 is proceeding passing it to Zap/32-1 2005-07-28 17:23:41 DEBUG[13997] chan_zap.c: Requested indication 15 on channel Zap/32-1 2005-07-28 17:23:41 DEBUG[13997] chan_zap.c: Received AST_CONTROL_PROCEEDING on Zap/32-1 regards frank ___ 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] Extension Lights Patch
Tom Hayden wrote: I've been using the extension lights on my polycoms before that patch, so I'm not sure what it fixed, but I've only seen the lights work on Polycoms and Snoms. Try using the hint priority and see if it works for your gxp2000, be sure to post your results! this gives you support for blinking leds, when the monitored phone is ringing and the ablility to pick that call up, when the phone supports 'INVITE/Replaces' by hitting the button beneath the blinking led. regards frank ___ 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] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
hi listmembers, please test my new patch to chan_sip.c which is to make call pickup on the snom phones (and maybe other phones that support 'INVITE/Replaces') work and make comments in the bugtracker http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs. this patch sipsubscribe-20050715.rev779.txt enables: * monitoring of other lines (using the 'hint' priority) - LED off when monitored phone idle - LED on when monitored phone busy - LED blinking when monitored phone ringing * display of caller id on monitoring phone * call pickup by pressing function key beneath the blinking led * corrected MWI LED functionality * corrected MWI button functionality * dialplan extension for MWI button settable * major code cleanup and some other things i don't remember. with this patch the snom phone will _THE_ phone for a receptionist - and for every phone usergroup that likes easy call pickup. please test it extensivly and comment it in the bugtracker as i think many of us have been long waiting for this functionality (and it's a real pain for me to keep it up to date as it is a real big patch) regards frank (xylome) ___ 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] call pickup with snom phones
hi, is there anybody who was able to setup call pickup with a snom phone? searching through the web brought up this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom section call pickup but this doesen't seem to work with current releases of the snom firmware (and looking through the patch of easywe it never worked very good at all) current snom firmware doesn't seem to send the required INVITE/REPLACE messages. any help is appreciated. regards frank ___ 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 wireless on site personal paging system
hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6 (and not EuroISDN). - is there anybody with experience on these pager devices? do they have a common standard? - does anybody know of a pager base station with an SIP interface? - does anybody know of a pager base station with an EuroISDN interface? what's your general advice on those paging systems? regards frank sautter ___ 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] experience with analog channel banks in E1 land
hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in E1 land (germany) we have 64kpbs per channel. most (affordable) channels banks are T1 (56kpbs per channel i assume). the questions are: - could the T1 channelbanks be connected to a TE405P with two channels in E1 mode (telco and hicom pbx) and two channels to the channel banks (i think yes, but just to be shure)? - will the faxmachines work (56kpbs-64kbps)? is asterisk translating this (btw. how do faxes work from europe to north america - the telcos have the same problem)? - which signalling protocoll will be used on the T1 side? is asterisk translating this correctly? - btw. where is the different bitrate coming from? is it 7bit T1 and 8bit E1 or 7kHz and 8kHz sample rate? regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and wireless on site personal paging system
hi patrick, Patrick schrieb: Did you try contacting the vendor of the base stations to see if they have a EuroISDN firmware update? My Eicon Diva Server BRI card supports the 1TR6 protocol. The firmware can be found here: ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/ Perhaps AVM supports 1TR6 too. yes, eicon diva server supports (we also have one here) but i was not able to load the capi drivers upon the 1TR6 stack?!? the next problem would be, that we need a isdn interface in NT mode, which is (to my knowledge) only possible with the cologne chip cards (junghanns / beronet). so i think we need an new solutions with the old wireless pagers. is there anybody who has experience with http://www.ascom.com/ws ? ___ 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: chan_capi, chan_misdn and chan_modem
hello armin, Armin Schindler schrieb: - cleanup in chan_capi.c (I noticed some errors) - add native bridging using CAPI Line-Interconnect this would be very nice I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... chan_capi currently supports receiving and sending of faxes utilizing the onboard DSPs of the eicon cards. please look for the neccessary patches at: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 regards frank sautter ___ 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: chan_capi, chan_misdn and chan_modem
Armin Schindler wrote: please look for the neccessary patches at: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 That is exactly what I was thinking about. I did not have a close look into the patch yet, but this archive seems to be incomplete. Only changed files are part of the patch, but the real app_capiFax.c is missing. The patch was obviously created without the '-N' option of diff. Can you please check that? you are right. i just updated the patch. it should work now. Anyway, I think this should be part of the chan_capi package. yes, i was very happy to hear from klaus-peter he has restarted to improve chan_capi (i thought he lost interest in chan_capi and concentrated only on his bri cards). i hope klaus-peter will include the fax support into chan_capi-0.4.0! So the CAPI on kernel 2.6 problem is on top now... fine. freundliche grüße frank sautter ___ 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, chan_misdn and chan_modem
hallo klaus-peter, Klaus-Peter Junghanns wrote: The new version of chan_capi (0.4.0) is still work in progress (no, I have not dropped chan_capi in favour of BRIstuff). that was my assumption, as there was no progress so many months. i'm very happy, that you are back on developing chan_capi! I harmonized the dialstring syntax with chan_zap, so you can just use CAPI/g1/... instead of those strange constructions with the outgoing msn. It also contains fixes (contributed by Jan Stocke) to make it work on BSD. Chan_capi 0.4.0 will work with Asterisk stable and cvs head. sounds as if this are interesting features. maybe you could take a look on the patches of carl sempla and cedrik hans (faxing with eicon cards) and mine (transfer capability, limitation of MSNs, cvs-head) both available using: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 grüße frank sautter ___ 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 (Digium E100P) problem : B-channel succesfully restarted.
hello jairo, Jairo Buendia wrote: -- B-channel 0/1 succesfully restarted on span 1 unused b-channels are reset by asterisk every hour (default). you can set the interval to another value in your /etc/asterisk/zapata.conf resetinterval=86400 ; e.g. reset every 24hours or even longer. i think since 2 months or so this should work also. resetinterval=never regards frank ___ 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] Debugging zaphfc + PBX integration
Gavin Hamill wrote: I know the cables themselves are wired correctly because our local PBX support made them, and they work perfectly when plugged into a real BT ISDN2e wallbox it seems as if this is exactly your problem. the wallbox has a NT pinout = straight trough cable the hfc card has a TE pinout = you need a cross-over (isdn not ethernet!!) cable to connect to your local pbx which also has a TE pinout. the nt/te switches on the hfc card do not cross over the rx/tx pairs of the card. this has to be done with the cabling. i don't think you will need any termination resistors if your cable is only a few meters and does not have any other devices on the bus. regards frank sautter ___ 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/T1 back to back ??
hi gary, Brett, Gary wrote: (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) you need a crossover cable (not the same as a ethernet x-over) take a look at: http://www.voip-info.org/wiki-crossover+T1+cable frank ___ 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 hardware for this solution?
hi giorgio, Giorgio Mandolfo schrieb: - an Asterisk machine connected to a traditional PBX (s0). In this way people is not (yet) obligated to migrate its extisting PBX (and analog phones) to VoIP. Straight to the point: what kind of hardware I need? I saw some PCI cards (like Digium Wildcard TE110P) but I am not sure what to buy. if you want to splice asterisk between a pbx and an S0 (ISDN-BRI) from your telco then you will need a ISDN Card that support NT mode e.g. cards with HFC chipset like those from www.junghanns.net or www.beronet.com. regards frank sautter ___ 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] Zaptel.conf and multiple T1 woes
Ben Ruset wrote: I have two Digium cards. One is a TE405P quad T1 card. The other is a TDM40B (I believe) quad analog POTS card. Our provider has been telling us that they are only seeing one D channel active. This would make sense if somehow only the first T1 in the 405P was activated. maybe it's a sync problem. i had trouble with a both the TE405P and a TDM40B in in the same system. somehow the ztconf or chan_zap is configuring the spans wrong if the kernel module for the TDM40B is loaded before the TE405P. lsmod shows the modules in reversed load order. set the sync source to span 1: span=1,1,0,esf,b8zs what are the effects you experience (besides there is no d-channel on one line)? regards frank sautter ___ 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] patch for chan_capi to compile with latest CVS
hi * users, due to the fundamental code changes in cvs tonight, it was necessary to update chan_capi to the new channel_tech design. it completely replaces my former patch from november 2004. the patch can be downloaded at http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 regards frank sautter ___ 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-PRI: Warning Message: Unable to handle ROSE operation 36
hi, since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? regards frank ___ 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] A: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
hi, i made a patch which allows the forwarding and the setting of the Bearer Capability ID during the ISDN SETUP phase. this solves several problems (primarily faxing) with SIN (german: Dienstekennung) and asterisk. http://bugs.digium.com/bug_view_page.php?bug_id=0003547 Frank Sautter wrote: i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. regards frank ___ 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] Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set the SIN so the called party is able which device has to answer a call (e.g. telephone or fax) as far as i dug into the source neither the BC nor the HLC or LLC data is forwarded to a dialplan variable and only the BC is decoded in libpri. has anyone a solution for this? is there any usable documentation on the HLC or LLC octets (bytes)? i searched etsi and was overwhelmed with the searchresults (1531). what i need to modify libpri would be a table of possible values and where to find the HLC and LLC fields in the D-Channel. regards frank ___ 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] Q: How to get the preset callerid from a CLID-no-screen E1-PRI
Peter Svensson wrote: This is rather weird? this are also my thoughts... What network do you receive this from? the calling party has an E1-PRI from the Deutsche Telekom (germany's former monopolist) and our E1-PRI is from Arcor which is on of the new telco companies founded after the liberation of the telco market in germany. as mentioned on my first email, they have enabled the ISDN feature CLIP-no screening but from the debug it seems, as if a screened number is sent after the unscreened number. but this is not the only caller i noticed this behaviour. a bank in our town also shows this behaviour (but arranging a call from them is not so easy). Neither ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the Calling Number IE among those that may be repeated. but it seems as if this behaviour is 'normal' because on every telephone line i tested this, the number shown is the unscreened number which came first in the debug I am quite certain that libpri does not handle this. The last one will overwrite the earlier calling numbers. that's the behaviour it currently shows. Some hacking of libpri is probably needed to handle this. To handle it cleanly a more complex interface between chan_zap and libpri may be needed. i think i will open a bug and dicuss how to implement this issue. i think both number are of interest, but the behaviour of the other telco equipment should be retained... showing the unscreened number. regards frank ___ 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] Q: How to get the preset callerid from a CLID-no-screen E1-PRI
Kevin P. Fleming wrote: Frank Sautter wrote: our customer uses this feature to show the callerid of the original caller when redirecting a call to a mobile phone. That is RDNIS, it shows the redirected number. In other words, it's not CLID (Calling Line ID). Check the RDNIS channel variable to see what it holds when you receive one of these calls. no, they are not deflecting the call. they are answering the call and making a new one to the mobile phone. RDNIS is empty. the main problem are not the redirected calls, but 'normal' calls from there showing the trunk CLID instead of the trunk CLID plus the local extension in the CLID. if anyone is interested, i can arrange calls from there for debugging purpose. the CLID shown should be +497031714717 but actually shown is +4970317145 (the assigned number range they have is +497031714500 to +497031714999) regards frank sautter ___ 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] Q: charge info on E1-PRI
hi, how can the charge info from a E1-PRI be received and be forwarded to a classic PBX? regards frank ___ 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] Q: how to receice the number of the called party back?
hi, a feature of euroisdn is, that you dail a number e.g. 0732194490 (where 0 is the extension of the call dispatcher) and the phone is forwarded to someone with an extension of 26. our ericsson showed after the call was picked up 07321944926 and no longer the dialled 0732194490. another example is, that you are dialling a number within the same area code e.g. 9876543 and after the phone is picked up the number in the display of the caller change to 07119876543. how can this be achieved? regards frank ___ 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] A: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid
Frank Sautter wrote: on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission based on the length of the presented callerid, as the length of the callerid can vary in most countries. e.g.: i'm getting signalled 4123456789 which could be a call from Barmstedt (Germany) which has the areacode '4123' or from Switzerland which has the countrycode '41' somehow our ericsson businessphone 250 fromerly connected to the same E1-PRI was capable of showing the correct number of leading 0s?!? the patch i made is now available through CVS-HEAD. thanks again to peter svensson who gave me the relevant hints where to look after! it is now possible to define prefixes in zapata.conf internationalprefix=00 nationalprefix=0 localprefix=089 privateprefix=0891234 unknownprefix= is also made the channel restart interval per span configurable resetinterval=86400 regards frank ___ 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] Q: How to get the preset callerid from a CLID-no-screen E1-PRI
Peter Svensson wrote: On Fri, 4 Feb 2005, Frank Sautter wrote: RDNIS is empty. So the operator sets an incomplete callerid? Sounds like a misconfiguration at the operators end. Do a pri intense debug span XXX on one of the calls and post the log of the SETUP to CONNECT_ACK messages. Protocol Discriminator: Q.931 (8) len=55 Call Ref: len= 2 (reference 4122/0x101A) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 21 80 31 37 32 39 38 37 36 35 34 33] 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 not screened (0) '1729876543' ] [6c 0a 21 83 37 30 33 31 37 31 34 35] Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '70317145' ] [70 08 c1 31 32 33 34 35 36 37] Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1234567' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) as you can see there are two calling numbers sent: 1729876543 (CLI-no screen - which should be the callerid) and a second one 70317145 (Network provided number). Are you sure that the RDNIS is empty? yes, here is a debug output of this call where i put out all variables i get -- Executing NoOp(Zap/1-1, -user2user -c2ton 65 -csub -ani2 00 -cton 33 -ctns 0 -cani2 0 -cnum 070317145 -id 070317145 -rdnis -pres 3) ___ 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] Q: How to get the preset callerid from a CLID-no-screen E1-PRI
hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a Caller ID no screen capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter if the callerid is within or without the assigned callerid range of the calling PBX. our customer uses this feature to show the callerid of the original caller when redircting a call to a mobile phone. i searched through the sourcecode of chan_zap and a little bit through libpri but did not find any variable containing the number. our ericsson BP250 PBX formerly connected to our E1 showed the callerid set by the caller correct. in the CDR of the ericsson both numbers appeared. any help is appreciated. regards frank sautter ___ 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] Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid - solved
hi, Frank Sautter wrote: on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. after peter svensson gave me some hints on where to look after, i made a small patch to current cvs-head which should solve this problem, by returning the modified callernum. the patch can be found on http://bugs.digium.com/bug_view_page.php?bug_id=0003493 regards frank ___ 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] AGI Script for CID Rewrite and CID Name lookup
hi jay, Jay Milk wrote: The result can be found here: http://www.muware.com/asterisk/ it seems as if your webserver tries to execute the .php file instead of making them available for download... regards frank ___ 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] Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid
hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission based on the length of the presented callerid, as the length of the callerid can vary in most countries. e.g.: i'm getting signalled 4123456789 which could be a call from Barmstedt (Germany) which has the areacode '4123' or from Switzerland which has the countrycode '41' somehow our ericsson businessphone 250 fromerly connected to the same E1-PRI was capable of showing the correct number of leading 0s?!? regards frank ___ 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] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
Frank Sautter schrieb: * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten = _.,102,Busy() with no effect. this is the part of the extensions.conf i'm using: peter svensson gave me the hint to set priindication=outofband now i'm able to signal busy to the calling party and with setting PRI_CAUSE there are even more possibilities see http://www.voip-info.org/wiki-Asterisk+cmd+Hangup regards frank ___ 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] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250
hi,
thanks to peter i solved my problems with the asterisk server spliced
between the telco and our ericsson BP250.
the problem was solved by setting 'overlapdial=yes'
Peter Svensson wrote:
Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter:
the setup desired with asterisk spliced in:
Arcor TelCo PRI(E1) P1 asterisk P2--- Ericsson BP250 PRI(E1)
Extension '' in context 'pri-ericsson' from '123498765' does not
exist
It sounds like the Ericsson pbx uses overlap dialing. Try enabling
that on both links in the zapata.conf file and see if it works better.
For immediate=no you should not match the s context. I think
exten = _.,1,Dial(Zap/g2/${EXTEN})
is more correct. Or use _XXX for a three digit DID.
i had to modify my dialplan on some points (thanks again to peter) and
twiddle with the callerid, our trunk MSN and the extensions, but it
seems to work.
today is our first working-day with asterisk in-between - so far no
problems (i hope it keeps this state).
here are the essential parts of the configuration files.
/etc/zaptel.conf
# TDM40B quad fxs analog-modules
span=1,0,0,ccs,hdb3,crc4
fxoks = 1-4
# TE405P/TE410P quad E1
span=2,1,0,ccs,hdb3,crc4
bchan=5-19,21-35
dchan=20
span=3,0,0,ccs,hdb3,crc4
bchan=36-50,52-66
dchan=51
span=4,2,0,ccs,hdb3,crc4
bchan=67-81,83-97
dchan=82
span=5,0,0,ccs,hdb3,crc4
bchan=98-112,114-128
dchan=113
loadzone=nl ; there is no 'de' zone right now
defaultzone=nl
/etc/asterisk/extensions.conf
[pri-external]
exten = _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero
exten = _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI
exten = h,HangUp()
include = durchwahl
include = pri-external-route
[pri-external-route]
exten = _.,1,Dial(Zap/g3/${EXTEN})
[pri-ericsson]
include = durchwahl
include = pri-ericsson-route
exten = h,HangUp()
[pri-ericsson-route]
exten = _XX.,1,SetCIDNum(${CALLERIDNUM:8})
exten = _XX.,2,SetCIDName('my name')
exten = _XX.,3,Dial(Zap/g2/${EXTEN})
/etc/asterisk/zapata.conf
[channels]
;### Quad FXS Card (TDM40B)
language=de
context=analog-lines
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
usecallingpres=yes
sendcalleridafter=1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling=fxo_ks
callerid=Harry Hirsch171
mailbox=171
accountcode=analog1
channel = 1
callerid=Hans Dampf172
mailbox=172
accountcode=analog2
channel = 2
callerid=Mork vom Ork173
mailbox=173
accountcode=analog3
channel = 3
callerid=Faxe179
mailbox=0
accountcode=analog4
channel = 4
;### Quad PRI(E1) Card (TP405P/TP410P)
language=de
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
musiconhold=default
callgroup=1
pickupgroup=1
immediate=no
overlapdial=yes
accountcode=pri
context=pri-external
group = 2
signalling=pri_cpe
channel = 5-19,21-35
context=pri-ericsson
group = 3
signalling=pri_net
channel = 36-50,52-66
context=pri-debug1
group = 4
signalling=pri_cpe
channel = 67-81,83-97
context=pri-debug2
group = 5
signalling=pri_net
channel = 98-112,114-128
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[Asterisk-Users] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi,
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten = _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
[pri-external]
exten = _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero
exten = _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI
exten = h,HangUp()
include = durchwahl
include = pri-external-route
[pri-external-route]
exten = _.,1,Dial(Zap/g3/${EXTEN})
exten = _.,2,Hangup()
exten = _.,102,Busy()
this is a excerpt from /var/log/asterisk/full
a call from a mobile phone (017212345678) to extension 134 which is busy:
-- Starting simple switch on 'Zap/35-1'
-- Executing SetCIDNum(Zap/35-1, 017212345678) in new stack
-- Executing Goto(Zap/35-1, 134|1) in new stack
-- Goto (pri-external,134,1)
-- Executing Dial(Zap/35-1, Zap/g3/134) in new stack
-- Called g3/134
-- Zap/38-1 is making progress passing it to Zap/35-1
Requested indication 14 on channel Zap/35-1
Received AST_CONTROL_PROGRESS on Zap/35-1
Dunno what to do with control type 15
-- Zap/38-1 is busy
Set option AUDIO MODE, value: ON(1) on Zap/38-1
Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 38
Set option TDD MODE, value: OFF(0) on Zap/38-1
Updated conferencing on 38, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/38-1
disabled echo cancellation on channel 38
-- Hungup 'Zap/38-1'
== Everyone is busy/congested at this time (1:1/0/0)
Exiting with DIALSTATUS=BUSY.
-- Executing Busy(Zap/35-1, ) in new stack
Requested indication 5 on channel Zap/35-1
== Spawn extension (pri-external, 134, 102) exited non-zero on 'Zap/35-1'
-- Executing Dial(Zap/35-1, Zap/g3/h) in new stack
-- Called g3/h
Set option AUDIO MODE, value: ON(1) on Zap/38-1
Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 38
Set option TDD MODE, value: OFF(0) on Zap/38-1
Updated conferencing on 38, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/38-1
disabled echo cancellation on channel 38
-- Hungup 'Zap/38-1'
Exiting with DIALSTATUS=CANCEL.
== Spawn extension (pri-external, h, 1) exited non-zero on 'Zap/35-1'
Set option AUDIO MODE, value: ON(1) on Zap/35-1
Hangup: channel: 35 index = 0, normal = 60, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 35
Set option TDD MODE, value: OFF(0) on Zap/35-1
Updated conferencing on 35, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/35-1
disabled echo cancellation on channel 35
-- Hungup 'Zap/35-1'
regards
frank sautter
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[Asterisk-Users] Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * from time to time (sometime within a few minutes sometime after hours) a complete PRI line or several PRI lines are kind of resetting (none of my colleagues reported a call interruption though). could this be a problem of the length (around 4kilometres) of the line between the telco switch and the NT providing the E1-PRI? The PRI line itself is only 3 metres long. is this the line build-out parameter in /etc/zaptel.conf? or is this something with timing of the span? my current settings are: # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # TE405P/TE410P quad E1 span=2,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 span=3,0,0,ccs,hdb3,crc4 bchan=36-50,52-66 dchan=51 span=4,2,0,ccs,hdb3,crc4 bchan=67-81,83-97 dchan=82 span=5,0,0,ccs,hdb3,crc4 bchan=98-112,114-128 dchan=113 this is a excerpt from /var/log/asterisk/full -- B-channel 0/1 successfully restarted on span 2 -- B-channel 0/3 successfully restarted on span 2 -- B-channel 0/5 successfully restarted on span 2 -- B-channel 0/6 successfully restarted on span 2 -- B-channel 0/7 successfully restarted on span 2 -- B-channel 0/8 successfully restarted on span 2 -- B-channel 0/9 successfully restarted on span 2 -- B-channel 0/10 successfully restarted on span 2 -- B-channel 0/11 successfully restarted on span 2 -- B-channel 0/12 successfully restarted on span 2 -- B-channel 0/13 successfully restarted on span 2 -- B-channel 0/14 successfully restarted on span 2 -- B-channel 0/17 successfully restarted on span 2 -- B-channel 0/18 successfully restarted on span 2 -- B-channel 0/19 successfully restarted on span 2 -- B-channel 0/20 successfully restarted on span 2 -- B-channel 0/21 successfully restarted on span 2 -- B-channel 0/22 successfully restarted on span 2 -- B-channel 0/23 successfully restarted on span 2 -- B-channel 0/24 successfully restarted on span 2 -- B-channel 0/25 successfully restarted on span 2 -- B-channel 0/26 successfully restarted on span 2 -- B-channel 0/27 successfully restarted on span 2 -- B-channel 0/28 successfully restarted on span 2 -- B-channel 0/29 successfully restarted on span 2 -- B-channel 0/30 successfully restarted on span 2 -- B-channel 0/31 successfully restarted on span 2 regards frank sautter ___ 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] analog fax on ericsson BP250 - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * some faxes from our analog fax-machine on our ericsson BP250 do not get through or only after several tries. regards frank sautter ___ 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 splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250
hi,
i'm having problems getting asterisk spliced between an E1 PRI (german
Telco Arcor) and an Ericsson Business Phone 250 digital PBX.
The Asterisk Server has a TE405P with it's port 1 connected to the E1
PRI provided by our telecommunications provider Arcor and port 2
connected to the E1 PRI of our Ericsson BP250.
the setup before:
Arcor TelCo PRI(E1) Ericsson BP250 PRI(E1)
the setup desired with asterisk spliced in:
Arcor TelCo PRI(E1) P1 asterisk P2--- Ericsson BP250 PRI(E1)
receiving and making calls between asterisk and the outside (arcor)
works so far (not entirely tested yet), but making calls from the
ericsson PBX to the asterisk server and routing them through to the
arcor PRI is not working.
the message i get when making a call from the ericsson pbx is:
Extension '' in context 'pri-ericsson' from '123498765' does not exist
obviously the ericsson pbx is not sending the dialled number on the pri
(but the calling number is set correctly)
as there is very limited time for me to play around with the parameters
in the asterisk config files (as the ericsson is in production use), i
hope the community can help me.
i think zaptel.conf is OK, as the LEDs are all green and the
communication between the all devices is working.
do i have to make changes on the ericsson PBX or in the zapata.conf?
regards
frank
here are some fragments of my config files:
/etc/zaptel.conf
# TE405P quad PRI(E1)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109
loadzone=nl
defaultzone=nl
/etc/asterisk/zapata.conf
[channels]
language=de
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.8
txgain=0.8
callgroup=1
pickupgroup=1
immediate=no
context=pri-external
group = 1
signalling=pri_cpe
channel = 1-15,17-31
context=pri-ericsson
group = 2
signalling=pri_net
channel = 32-46,48-62
context=pri-loopin
group = 3
signalling=pri_cpe
channel = 63-77,79-93
context=pri-loopout
group = 4
signalling=pri_net
channel = 94-108,110-124
/etc/asterisk/extensions.conf (just the part of it that matters)
[pri-external] ; calls from the telco
include = durchwahl
exten = s,1,Answer()
exten = s,2,Dial(Zap/g2/${EXTEN})
exten = s,3,Hangup()
[pri-ericsson] ; calls from the ericsson BP250 to asterisk
include = durchwahl
exten = s,1,Answer()
exten = s,2,DigitTimeout,2
exten = s,3,ResponseTimeout,10
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
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[Asterisk-Users] german dialtones for IAXy?
hi, is there a possibility to provide german dialtones on an IAXy S100IPWRD? 'language=de' sets only the messages to german (voicemail, etc.) is there something like 'loadzone' as in /etc/zaptel.conf regards frank ___ 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-0.3.5 error 127
hi vincent, Vincent Guidoux schrieb: I have a problem for install chan_capi My pc: Suse 9.1, with asterisk current stable en cvs And patch the chan_capi chan_capi.c:1076: error: structure has no member named cid as you are writing and apparent to the error message you are posting, you are using a stable 1.0.x version of asterisk. therefore you don't need to apply my patch, which is only for the HEAD-cvs version of asterisk. the version 0.3.5 from junghanns.net will (hopefuly) compile fine with your stable version. regards frank ___ 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] voicemail sound distorted - chan_capi, diva, cvs-head
hi, i have a problem with distorted voicemail sound on our asterisk test machine. i'm using cvs-head (2004-01-17) and chan_capi 0.3.5 (with my patches to make chan_capi compile with asterisk cvs-head) and a diva quad-bri isdn card. other things work well with my setup (dial in, dial out, app_meetme) and sound recordings from sip channels. the problem is with voicemail and app_record, where only a distorted sound can be heard in the recording if one shouts into the microphone of the telephone. has anybody had the same problem or can confirm this issue? regards frank ___ 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 outgoing msn
Vincent Guidoux schrieb: Now i have a un new prob Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in new stack Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request: didn't find capi device with outgoing msn = 4202270. you should check your config well the error message says it all. 'you should check your config' apparently you haven't configured your MSNs in /etc/asterisk/capi.conf. 8 snip [interfaces] msn=4202270 incomingmsn=* 8 snap regards frank ___ 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] include and hint in extensions.conf with new realtime feature - how?
hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and normal extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten = 183,hint,SIP/snom220 exten = 183,1,Macro(stdexten,443,SIP/snom220,183) exten = 187,hint,SIP/zyx2000 exten = 187,1,Macro(stdexten,447,SIP/zyx2000,187) the entries in the table for realtime config look like this: SELECT context, exten AS ext, priority AS prio, app, appdata FROM extensions WHERE exten IN (183,187); +--+-+--+-+--+ | context | ext | prio | app | appdata | +--+-+--+-+--+ | from-sip | 183 | hint | SIP/snom220 | NULL | | from-sip | 183 | 1| Macro | stdexten|442|SIP/snom220|183 | | from-sip | 187 | hint | SIP/zyx2000 | NULL | | from-sip | 187 | 1| Macro | stdexten|443|SIP/zyx2000|187 | +--+-+--+-+--+ i also tried to put the parameter for the hint command into the 'appdata' column. my other problem is: how can includes of other contexts be accomplished? regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom - blinking leds on fuction keys when call is not yet established - how?
hi, i just ported the patch of David Hinkle http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html to the current cvs version of asterisk. the theory is that the leds of supervisioned extensions are blinking until a call is established whereafter the leds should be constantly lit. however it's not working. the asterisk server is sending the following xml notify to the snom phone (according to the sip trace of the phone) when the extension is ringing: dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=11 state=full entity=sip:[EMAIL PROTECTED] dialog id=185 stateearly/state /dialog /dialog-info the corresponding led is lit contantly and not blinking as it should be. the system information of the snom phone is as follows: Phone Type: snom220-SIP Version-Code: snom220-SIP 3.52 Bootloader: http://www.snom.com/download/snom220-boot2.1.bin Firmware: http://www.snom.com/download/snom220-3.52-beta-SIP.bin he functions keys are configured: fkey5!: dest sip:[EMAIL PROTECTED];user=phone how can the blinking state of the leds be achieved? is this a firmware version issue of the 3.52 i'm using? regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] patch for chan_capi to compile with latest CVS
hi john, John Williams schrieb: i made a patch that allows the compilation of chan_capi-0.3.5 against current CVS-HEAD of asterisk. If I remove the -2.95 from the CC declaration I get a very large number of errors, the same ones I get when trying to compile without the patch. ok, i forgot to change this back to gcc. i just changed it in my local sources and it also compiles without a problem with a gcc version 3.4.2. have you done 'make install' to the asterisk cvs sources *first*? chan_capi includes some header files from asterisk which are installed in /usr/include/asterisk/. alternatively you could change the include paths of chan_capi pointing to your asterisk source tree. if the problem persists you can send me the output of your compiler and i could check where the problem is. greetings to the other side of the globe frank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] patch for chan_capi to compile with latest CVS
hi, i made a patch that allows the compilation of chan_capi-0.3.5 against current CVS-HEAD of asterisk. it also incorporates the capiAnswerFax patch the patch can be downloaded at http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 regards Frank Sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk support for ISDN 1TR6 ?
hi, can someone give me any hints if the old german ISDN protocol '1TR6' is supported by asterisk. we have a potential customer who has an existing conventional PBX which has to be extended by an asterisk server. unfortunately this existing PBX speaks 1TR6 on it's ISDN ports. regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk support for ISDN 1TR6 ?
well i have an icon diva quadbri card and i already tried uploaded the 1TR6 firmware, which seems to work so far. the problem is, that the capi module and therefore chan_capi do not load correctly. Patrick wrote: I know my Eicon Diva Server BRI card supports 1TR6 on the ISDN side and works fine with Asterisk. To activate 1TR6 all I would have to do is upload the proper firmware to the card. Maybe the AVM Fritz! cards support 1TR6 too. Worth checking out. The Eicon cards are expensive while the AVM Fritz! is much cheaper. On Thu, 2004-11-11 at 11:38 +0100, Frank Sautter wrote: can someone give me any hints if the old german ISDN protocol '1TR6' is supported by asterisk. we have a potential customer who has an existing conventional PBX which has to be extended by an asterisk server. unfortunately this existing PBX speaks 1TR6 on it's ISDN ports. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] german patches for say.c
hello, i just wanted to inform you, that i made some patches to say.c so * can speak numbers and dates in a correct german syntax. the patches are available through http://bugs.digium.com/bug_view_page.php?bug_id=0002780 a compatible (but not complete) set of german sounds can be found on http://www.stadt-pforzheim.de/asterisk/ there will also be a more complete set of german sounds, but currently the female voice has a cold so it will take a few days until they are available. regards, frank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] searching for a nifty solution for different outgoing msn depending on the sip-user
hi,
our asterisk server is currently connected via 4 isdn trunks to our main
pbx using it as a voip gateway for homeworkers.
currently this is the dial command for outgoing calls
exten = _., 1, Dial,CAPI/141:${EXTEN}
what i like to do, is giving each sip-user a different outgoing msn (the
141 in the example above).
the only solution i found, is to put each user into a different context,
but this leads to a very complex and error-prone extentions.conf.
is there a niftier solution?
thanks
frank
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[Asterisk-Users] german localization for mailbox available?
hi, i just wanted to ask if there is a german localization for the audio files of the mailbox available on the net. regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
