Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service
On Thu, 17 Nov 2005, Frederic Steinfels wrote: Last January I told KPJ that I can still not use my Simens Gigagaset cordless phones and sent him some bug reports. He promised me to fix this bug several times but nothing happened. The problem is that the phone is displaying the word Störung (english probably out of order) within days of using it requiring modules to be unloaded/loaded and asterisk restarted. Are you sure the problem isn't on the Siemens' side? I have a Siemens Gigaset SX303isdn connected to my asterisk (bristuff-0.2.0-RC8p) and the Gigaset keeps running out of TEIs after some while (since TEI:s lost by the terminal equipment will not be reused until asterisk is restarted). Scheduled asterisk restarts every week seems to solve the problem. With other ISDN phones this problem does not occur. -- Regards, Tobias Jönsson, Lund SE___ --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] zaphfc syslog flooding
On Tue, 9 Aug 2005, Arik Funke wrote: my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this normal?: -- zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353, wanted 8 got 7), probably a buffer overrun. zaphfc: dropped audio (z1=2712, z2=2695, wanted 8 got 17, dropped 9). Try googling these error messages and you'll get a lot of hints. Here's my suggestions anyway: 1. Are you using IDE disks on your server? You shouldn't, or at least you should set hdparm -u1 /dev/hd?. 2. cat /proc/interrupts to see if the zaphfc card is sharing irq with any other card. It shouldn't. If your motherboard supports IO-APIC you could try enabling it, otherwise try a different PCI slot. 3. Try removing other hardware that might require too much time during an interrupt. For example I had a r8169 network card that caused a buffer overflow every 10 seconds if the Ethernet cable was disconnected. 4. Use a current bristuff version (currently 0.2.0-RC8n). If you have several zaphfc cards in the same machine, try out Florz's patches http://zaphfc.florz.dyndns.org/. (Always get only one hfc card working before adding more.) 5. Your hardware might have problems with the many interrupts the zaphfc card generates. Some motherboards simply cannot take care of the 8000 interrupts per second these cards generate. Good luck! -- Best Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.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] Time sync on PRI
On Thu, 31 Mar 2005, Peter Svensson wrote: It would not be very hard to add both features to libpri. Libpri already has a function to decode and dump the time/date information. If I remember correctly the time/date IE should be added to the SETUP messages. I have been thinking about adding it, but have not had the time. It's already there, in bristuff patches. Please encourage Digium to add Junghanns' patches to the asterisk code :) -- Best Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and echocancel
mattf skrev: I checked many sample configs and the archives and noticed that half of the people have echocancel on for PRIs and half do not. I checked the Digium site and indeed in the FAQ they say: There should also be no echo on PRI connections. If you need echo cancelling on the PBX PRI depends on the phones used on that PBX. If they are all digital and they do not introduce any echo you will not need any echo cancelling. On the T1 side you will probably need echo cancelling since there might be an analog phone in the other end. If you only use PRI in and out on the same asterisk server the delays may be so short on a bridged call that the echo would not be noticable even if it is there. That is the way ordinary pstn calls work - there is usually echo cancellation only on cellular calls or long distance calls. -- Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and echocancel
Eric Wieling wrote: Tobias Jönsson wrote: If you need echo cancelling on the PBX PRI depends on the phones used on that PBX. If they are all digital and they do not introduce any echo you will not need any echo cancelling. The outside phone will still (usually) be analog and cause echo. So unless you are only calling other VoIP or cell phones you still need echo canceling when using PRI. Yes, that is exactly what I wrote??: On the T1 side you will probably need echo cancelling since there might be an analog phone in the other end. -- Best Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Bristuff ZapHFC and Loosing D-Channel
On Thu, 27 Jan 2005, Klaus-Peter Junghanns wrote: Am Donnerstag, den 27.01.2005, 16:01 +0100 schrieb Remco Barende: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up that is the usual behaviour on a P2MP BRI line. When idle the telco will bring down layer 2 and layer 1. Bristuff will activate layer 1 and layer 2 again immediately. What will happen if there comes a call (from asterisk or from ISDN network) during the short times when D-Channel is down? Will they retry or will the call be dropped? -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.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-tone on incoming calls?
On Tue, 25 Jan 2005, Eric Wieling wrote:
You can set a PRI_CAUSE variable. See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20PRI_CAUSE
This only works in CVS-HEAD. For production use just run Busy() in the
dialplan.
No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0
releases too. Busy() may play a busy tone to the caller instead of
signalling busy so using PRI_CAUSE is much better in PRI or BRI
environment.
exten = 123437,1,Dial(Zap/g2/37,26,tg)
exten = 123437,2,GotoIf($[${DIALSTATUS} = BUSY]?110:3)
exten = 123437,3,Answer
exten = 123437,4,Wait(1)
exten = 123437,5,Voicemail(su21)
exten = 123437,6,Hangup
exten = 123437,110,SetVar(PRI_CAUSE=17)
exten = 123437,111,Hangup
--
Regards,
Tobias Jönsson, Lund SE___
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Re: [Asterisk-Users] BUSY-tone on incoming calls?
On Tue, 25 Jan 2005, Peter Svensson wrote: On Tue, 25 Jan 2005, Tobias Jönsson wrote: No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0 releases too. Busy() may play a busy tone to the caller instead of signalling busy so using PRI_CAUSE is much better in PRI or BRI environment. The behaviour of Busy() and Congestion() can be changed with the priindication setting in zapata.conf. The options are inband (default) or outofband. This only affects the two applications mentioned above. Thank you for that information. I have now updated the wiki of zapata.conf. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.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] Meetme conf and Shoutcast
On Sun, 16 Jan 2005, Mike wrote: We would like to know if there is a way to broadcast (in realtime) a conferance. http://www.voip-info.org/wiki-Asterisk+cmd+Ices I haven't tried it though. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cannot open /dev/dsp
On Wed, 24 Nov 2004, Norman Zhang wrote: Cannot open /dev/dsp: file or directory not found You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. You do not need any sound card if you don't want to use the console channel drivers. Just take a look at your /etc/asterisk/modules.conf and be sure not to load them (noload = chan_oss.so, noload = chan_alsa.so). -- Regards, Tobias Jönsson, Lund SE___ 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] Authenticate or DISA?
On Tue, 16 Nov 2004, Peter Svensson wrote: On Tue, 16 Nov 2004, Tobias Jönsson wrote: On Mon, 15 Nov 2004, Jason Williams wrote: After the Authenticte why not do a Playtones(Dial) this will give dialtone The dialtone won't stop after pressing first digit then. If course you can have an X extension that will do a StopPlaytones but that is not a good solution since that one cannot be used for extension matching in further contexts. I missed the start of the thread. What is wrong with using the Disa application? DISA doesn't use the ResponseTimeout and DigitTimeout from extension logic. Earlier versions of DISA did also not use indication tones from indications.conf, but that seems to be corrected lately. -- Regards, Tobias Jönsson, Lund SE___ 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] Authenticate or DISA?
On Mon, 15 Nov 2004, Jason Williams wrote: After the Authenticte why not do a Playtones(Dial) this will give dialtone The dialtone won't stop after pressing first digit then. If course you can have an X extension that will do a StopPlaytones but that is not a good solution since that one cannot be used for extension matching in further contexts. -- Regards, Tobias Jönsson, Lund SE___ 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] CHANUNAVAIL = CHANUNAVAIL doesn't eval properly
On Fri, 15 Oct 2004, Matthew Boehm wrote: What 'should' happen is that if someone dials any extension starting with 3, Dial attempts to dial it. If there is no such channel, set DIALSTATUS and goto priority n + 101. Then check result of DIALSTATUS. If DIALSTATUS is equal to CHANUNAVAIL then goto extension 'i' else goto priority 103. Why don't you just use the bristuff package? There is a Dial application which goes to n+201 if channel is unavailable. -- Regards, Tobias Jönsson, Lund SE___ 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] No audio on PRI channel answered by Playback()orMeetMe()
On Fri, 27 Aug 2004, Larry Shields wrote: Thanks for the reply. I tried that initially and it did not work. To verify I went back and tried again. It answers and still no sound is heard. -- Accepting call from '8541' to '2688' on channel 0/2, span 1 -- Executing Wait(Zap/2-1, 3) in new stack -- Executing Answer(Zap/2-1, ) in new stack Why do you start with a Wait statement? Just answer the line immediately if you want to do that, or you should at least put a Ringing before the first wait statement if you want the caller to hear a ringing tone before you answer. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compressing a dialplan
On Wed, 25 Aug 2004, Maron Kristófersson wrote: Hmm, that raises a lot of questions for the script... How many contexts do you have? Do they include each other. Is there any kind of rule around the extensions... etc. All the extensions will just run the same macro so all these are just the same. What I did is that I converted the Swedish national numbering plan (E.164) with an AWK script to an extension like file. The purpose is being able to use early dialling for national calls (no overlap dialling available). It contains the starting digits and number lengths, like this: 04623 0462400XXX 0462401XXX 0462402XXX 0462403XXX 0462404XXX 0462405XXX 0462406XXX 0462407XXX 0462408XXX 0462409XXX 046241 046242 046243 046244 0462450XX 0462451XX 0462452XX 0462453XX 0462454XX 0462455XX 0462456XX 0462457XX 0462458XX 0462459XX 046246XXX 046247XXX 046248XXX 046249XXX 04625 0462600XXX 0462601XXX 0462602XXX 0462603XXX 0462604XXX 0462605XXX 0462606XXX 0462607XXX 0462608XXX 0462609XXX 046261 046262 046263 046264 046265 046266 046267 046268 046269 It's obvious that the following four lines will match exactly the same numbers as all the lines above: 0462[35] 04624[0-4] 04624[5-9]XXX 04626X That is the conversation I would like a macro/script to do. What I thought about was if there are any kind of regexp compression programs or something like that. -- Regards, Tobias Jönsson, Lund SE___ 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] GrandStream HT-486 ATA as VoIP Gateway
On Wed, 25 Aug 2004, Miroslav Nachev wrote: Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? I have one and it works quite well. Its echo cancelling is too good :), resulting in a feel of half duplex audio, and the router thrughput is aweful (they are just 10 Mbps Ethernet ports and the thrughput is about 2 Mbps). There are some bugs that the grandstream people do not seem to care about, for example you cannot send DTMF tones when you receive a call - only when you are the caller. It also just provides American indication tones which could be quite confusing to europeans. -- Regards, Tobias Jönsson, Lund SE___ 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] Compressing a dialplan
Does anyone have a program that could be used to compress the dialplan? I have lots of numbers in a list, for example if the file contains 12300 12310 123113 12320 12330 12340 12350 12360 12370 12380 12390 they all could have been written as two entries (123X0 and 123113) instead. Can anyone give me a hint please? Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Compressing a dialplan
On Wed, 25 Aug 2004, Maron Kristófersson wrote: Tobias Jönsson wrote: Does anyone have a program that could be used to compress the dialplan? I have lots of numbers in a list, for example if the file contains 12300 12310 123113 12320 12330 12340 12350 12360 12370 12380 12390 they all could have been written as two entries (123X0 and 123113) instead. Have you looked at http://voip-info.org/wiki-Asterisk+Dialplan+Patterns, The extensions _123X0 would handle all the extensions below except for the six digit extension. Yes exactly, that is why I ask for a program (or script or anything) that could do this transcription. My extensions.conf is about 6 lines so I cannot do it manually. Regards, Tobias Jönsson, Lund SE___ 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] Finding operator from ISDN signalling?
On Fri, 20 Aug 2004, Roy Sigurd Karlsbakk wrote: Is it possible to find out what source/destination operator you're connected to from the ISDN signalling? Number lookups no-longer work since number porting begun, and access to the national database for these numbers cost too much :( That information is, unfortunately, not included in signalling. In Sweden you can get the information for free (restricted number of queries) by a web service at Swedish Number Portability Administrative Center. I do not know if there is somethink similar in Norway. Regards, Tobias Jönsson, Lund SE ___ 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] call-back example
On Wed, 18 Aug 2004, Maros RAJNOCH wrote:
can anyone show me a exemple config for call-back?
1) I call asterisk server from my cellular
2) asterisk hang up my call (on d-channel)
3) asterisk recall to my cellular and give me a PSTN tone, so
I can to pick up a call and to dial new phone number (via tone dialing)
I already do that, so here you are!
[remote]
exten = yourmsn/yourcellphoneno,1,Goto(callback,${CALLERIDNUM},1)
[intern]
exten = 0,1,Dial(Zap/g1) ; I use overlap dialing at Zap/g1 but of course you could
collect some digits by yourself or with DISA application
[callback]
exten = _X.,1,SetVar(callbacknr=${EXTEN})
exten = _X.,2,SetVar(PRI_CAUSE=16)
exten = _X.,3,Hangup
exten = h,1,Wait(5); try out your delay needed
exten = h,2,System(echo Channel: SIP/[EMAIL PROTECTED] /tmp/${UNIQUEID}.call)
exten = h,3,System(echo MaxRetries: 2 /tmp/${UNIQUEID}.call)
exten = h,4,System(echo RetryTime: 60 /tmp/${UNIQUEID}.call)
exten = h,5,System(echo WaitTime: 30 /tmp/${UNIQUEID}.call)
exten = h,6,System(echo Context: intern /tmp/${UNIQUEID}.call)
exten = h,7,System(echo Extension: 0 /tmp/${UNIQUEID}.call)
exten = h,8,System(echo Priority: 1 /tmp/${UNIQUEID}.call)
exten = h,9,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing)
Regards,
Tobias Jönsson, Lund SE
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Re: [Asterisk-Users] Zaptel module loading (was: Another small suggestion patch)
On Wed, 18 Aug 2004, John Morris wrote: It's nice to be able to define the list of asterisk modules we want to load from the /etc/sysconfig/zaptel file rather than directly in /etc/init.d/zaptel. I'm using nufone and don't require anything but the ztdummy (is the rtc-based module better, anyone?), so that's what I've put here. Why don't you just add alias char-major-196 ztdummy to your /etc/modules.conf (modprobe.conf in linux 2.6)? Then your zaptel and ztdummy will be loaded when asterisk asks for them and you would not have to bother about loading them manually nor in your init scripts. Regards, Tobias Jönsson, Lund SE ___ 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] call-back example
On Thu, 19 Aug 2004, Peter Svensson wrote:
On Thu, 19 Aug 2004, Tobias Jönsson wrote:
exten = h,2,System(echo Channel: SIP/[EMAIL PROTECTED] /tmp/${UNIQUEID}.call)
exten = h,3,System(echo MaxRetries: 2 /tmp/${UNIQUEID}.call)
exten = h,4,System(echo RetryTime: 60 /tmp/${UNIQUEID}.call)
exten = h,5,System(echo WaitTime: 30 /tmp/${UNIQUEID}.call)
exten = h,6,System(echo Context: intern /tmp/${UNIQUEID}.call)
exten = h,7,System(echo Extension: 0 /tmp/${UNIQUEID}.call)
exten = h,8,System(echo Priority: 1 /tmp/${UNIQUEID}.call)
exten = h,9,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing)
There is a race in this solution of /tmp and
/var/spool/asterisk/outgoing are on different file systems. Then the
rename operation (mv) is not atomic but rather a copy.
Thank you Peter for pointing out that! My mistake. Unfortunately Asterisk
reads all files in the outgoing directory, not only *.call files, so it is
difficult to give a general solution to this. You can never be sure that a
file in a different directory is on the same file system, can you? Perhaps
the safiest would be to create a /var/spool/asterisk/outgoing/tmp
directory?
Regards,
Tobias Jönsson, Lund SE
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[Asterisk-Users] Re: Pingtel registration failing
On Thu, 19 Aug 2004, Anton Yurchenko wrote: The Asterisk sends the replies to port 1031, the outbound port that Pingtel used to send the message. This is wrong. In the REGISTER, Pingtel specified a contact header field with no port, which means use a default port of 5060. Asterisk is violating the RFC 3261 by ignoring the Contact header. As far as I understand, Asterisk will always ignore addresses in the SIP header if nat is enabled in sip.conf. Change the setting to nat=no and asterisk should follow the standard. Regards, Tobias Jönsson, Lund SE ___ 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] disable console channels
On Mon, 16 Aug 2004, Michael George wrote: When I start up *, though, it grabs my sound card and I cannot play other music through it (e.g. x/ XMMS). I have moved the alsa.conf and oss.conf files so that there is no configuration for them (though those files seemed to do little), but still the sound card is grabbed. How can I disable those channels? Just put these lines in your modules.conf: noload = chan_alsa.so noload = chan_oss.so Regards, Tobias Jönsson, Lund SE ___ 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] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004, Peter Svensson wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Not necessarily! When you call an analog phone via isdn, the other end will introduce echo so that the ip side will be hearing himself speaking with a small delay. I have that problem with my home BRI running zaphfc. Regards, Tobias Jönsson, Lund SE ___ 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] No incoming audio on incoming SIP calls
On Thu, 5 Aug 2004, John Howard wrote: Can anyone suggest anything as to why I still can't get call transferring working with the Zyxel 2000w wireless phones? Unfortunately not, but take a look into the ZyXEL 2000W threads in this forum, for example http://lists.digium.com/pipermail/asterisk-users/2004-July/055598.html and http://lists.digium.com/pipermail/asterisk-users/2004-June/051930.html. BTW, watch out for the difference between t and T options. Regards, Tobias Jönsson, Lund, Sweden ___ 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] zaphfc hardware sound trouble
On Fri, 30 Jul 2004 [EMAIL PROTECTED] wrote: ###zapata.conf context=default context=alex pridialplan=unknown echocancel=yes echocancel=yes echocancelwhenbridged=yes immediate=yes Why do you define echocancel and context twice? a) when i try to make an inbound call to msn I get the following message on the cli prompt -- Going to extension s|1 because of immediate=yes -- Extension 's' in context 'alex' from '17109904' does not exist. Rejecting call on channel 2, span 1 You should not use immediate=yes for a TE interface since that instructs asterisk to go to s extension (which is useful for NT interfaces but not for TE ones). I have set pridialplan=local and have the msn in exten = msn,... set including the area code. To only match calls from my cellular phone I use exten = msn/cellphonenumber,... That works fine for me. If it still does not work, turn on some debugging and try to catch what is happening when a call comes. b) the combination of my configuration with zaphfc and the acer pci isdn card seems to cause some other trouble Kernel: 2.4.21-0.13mdk Jul 29 23:46:49 faar kernel: sync lost, pci performance too low!!!. I had that problem running RedHat Linux 9.0 with kernel 2.4.20-31.9. It was probably occured by a buggy driver for the IDE controller integrated on my mainboard, that made the hard disk access interrupts taking too much time from the pci bus. The problem disappeared after upgrading kernel to 2.4.26. d) as long as the line works, I have clearly audible clicks/cracks in the line (zaphfc) that didn't occur using capi and the avm fritz! pci 2.0 - I don't have any sound problems on iax via voiptel.org or internally using sip This problem I still have. This is what I have found out: First turn with echocancel=no in zapata.conf. The echo cancelling does not work any good at all if some audio data is missing. That reintroduces echo while talking to analog phones, of course, but could help finding the source of your problem. Then check that your zaptel timing is all right. Try running zttest (in the zaptel directory) and watch the output. I still have problems with this one running zaphfc-0.1.0-RC2k on my new Intel Pentium 4 2,4 GHz system but timing is excellent on my old AMD K6-233 MHz system... Regards from Sweden, Tobias Jönsson ___ 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] zttest never get 100% accurancy
I never get 100% accurancy with zttest. Not even running ztdummy as timing source. Should it really be like that? Opened pseudo zap interface, measuring accuracy... 98.950195% 99.975586% 99.975586% 99.987793% 99.963379% 99.536133% 99.975586% 99.975586% 99.987793% 99.548340% 99.975586% 99.975586% 99.975586% 99.987793% 99.548340% 99.975586% 99.987793% 99.963379% 99.987793% 99.548340% 99.975586% 99.975586% 99.975586% 99.987793% 99.548340% 99.975586% 99.975586% 99.975586% 99.987793% 99.548340% 99.987793% 99.975586% 99.975586% 99.987793% 99.157715% 99.975586% 99.975586% 99.975586% 99.987793% 99.548340% --- Results after 40 passes --- Best: 99.987793 -- Worst: 98.950195 Regards, Tobias Jönsson ___ 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] outgoing MSN on zaphfc
On Wed, 26 May 2004, Klaus-Peter Junghanns wrote: exten = _X.,1,SetCallerID(MyMSN) To get this work with Telia ISDN, Sweden, I had to send the area code without 0 in the beginning, for example my MSN is 046-370544 but I had to SetCallerID(46370544) -- neither 370544 or 046370544 worked. Perhaps I should add I have pridialplan=local; I couldn't dial numbers like 9545046370544 (9545 for choosing service provider) when using pridialplan=national. Regards, Tobias Jönsson ___ 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] zaphfc: All DTMF tones are doubled
When I use signalling=bri_net_ptmp, immediate=no I can dial an extension, but after connect every dtmf is detected twice. With immediate=yes the dtmf are detected twice even when dialling the extension number. Could this be because the telephone adapter both sends INFORMATION and the inband dtmf could be heard by the zaptel engine? Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 100/0x64) (Originator) Message type: INFORMATION (123) Keypad Facility (len= 3) [ Keypad Facility (len= 3) [ 1 Keypad Facility (len= 3) [ 1 ] -- Processing IE 44 (Keypad Facility) [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/5-1] [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/5-1] Regards, Tobias Jönsson ___ 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] Rejecting Calls (SIT Tone/Invalid) Across PRI
On Wed, 26 May 2004, Steven Sokol wrote: I have a client who wants to allow callers to dial a DID which connects over a PRI to Asterisk. Asterisk will be analyzing the ANI data from each call to that DID and if it recognizes the ANI, it needs to effectively return an Invalid Number or Not Found some-such message across the PRI to prevent the user from being billed. I have been looking for the same thing but chan_zap seems to be programmed to acknowledge all incoming calls that there is a matching entry in extension.conf for, which means ringing tones will be given before starting the execution of application commands. But if there is no entry for a call it will be rejected. For example: exten = 123456/456789,1,Hangup exten = 123456/789012,1,Hangup Calls from 456789 or 789012 will be hangup after one ringing tone (you could run a AGI script here). Calls from other numbers will be rejected immediately. Regards, Tobias Jönsson ___ 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] D-Channel on span 1 up/down + frame slips with zaptelBRI
I have installed two HFC PCI A-cards running zaphfc from bristuff-0.0.2, which seems to work quite fine, but I continously receive the messages D-Channel on span 1 up followed by D-Channel on span 1 down with a few seconds interval. Why is that? Bri intense debug log and configuration files below. I don't need ztdummy or zaprtc, do I? I get some frame slips as well, but zaphfc should be enough to get the timing? Regards, Tobias T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (0) [ 00 d9 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 108EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter paulus*CLI [ 00 d9 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 108EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter paulus*CLI paulus*CLI paulus*CLI [ 02 d9 53 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 108EA: 1 M3: 2 P/F: 1 M2: 0 11: 3 [ DISC (disconnect) ] 0 bytes of data -- Got Disconnect from peer. Sending Unnumbered Acknowledgement [ 02 d9 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 108EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter Sending Set Asynchronous Balanced Mode Extended [ 00 d9 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 108EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended ) ] 0 bytes of data == D-Channel on span 1 down paulus*CLI [ 00 d9 73 ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 108EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Got UA from network peer Link up. -- Restarting T203 counter == D-Channel on span 1 up /etc/zaptel.conf: ; By the way, how do I know if I should use CAS or CCS resp. AMI or HDB3? loadzone=nl defaultzone=nl span=1,1,5,ccs,ami bchan=1-2 dchan=3 span=2,0,5,ccs,ami bchan=4-5 dchan=6 /etc/asterisk/zapata.conf: [channels] switchtype = euroisdn ; to/from ISDN PtMP signalling = bri_cpe_ptmp pridialplan=local echocancel=no immediate=no language=se group = 1 context=remote channel = 1-2 ; to/from the PBX signalling = bri_net_ptmp pridialplan=local echocancel=no ;overlapdial=yes immediate=no language=se group = 2 context=analog-ankn channel = 4-5 ___ 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
