Re: [Asterisk-Users] VoIP Cheap Asterisk
Have you tried without reinviting ?? (canreinvite=no) Is your * box behind a nat ? maxx Scott Pinhorne ha scritto: Hi All I have setup my asteriks to use voipcheap.com for the outgoing trunk on local calls (because they are free), my setup is below: register = username:[EMAIL PROTECTED] [voipcheap] type=peer host=sip.voipcheap.com domain=voipcheap.com dtmfmode=inband context=mycontext allow=all canreinvite=yes qualify=yes username=username password=password When I start asterisk I am able to make calls out via this trunk but only for a certain period (random) and then after this I get a 503 Forbidden error, if i restart asterisk then it connects and it is ok again for a certain period. The logs show: Forbidden - wrong password on authentication for INVITE but how can this be if I am able to make calls for a while before hand, I have tried playing with various settings but cannot get it constant, if anyone has any ideas then I would be very grateful as it is doing my head in now :-) Many thanks Scott Pinhorne VoxIT.co.uk ___ --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 ___ --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] Disabling zap echo cancellor from dialplan
Koopmann, Jan-Peter ha scritto: On Thursday, January 19, 2006 6:53 PM Massimo De Nadal wrote: If you are using bristuff add the m option like this: Dial(Zap/g1m/numbertodial) This will turn off the modulation (rxgain/txgain) and should (!) turn of the echo cancellation but I am not sure about this. Regards, JP Thank you very much. Option m solved my problems. maxx ___ --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
[Asterisk-Users] Disabling zap echo cancellor from dialplan
Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx ___ --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] TDM2400
Guillermo Salas M ha scritto: Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. ___ --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
[Asterisk-Users] iaxmodem through zaphfc
Hi Guys, I'm trying to send and receive some faxes using iaxmodem and hylafax through an hfc isdn board and a bristuffed asterisk. All seems to work fine, but the faxes are sent randomly truncated without any reported error. Any idea or suggestion ? Anybody owns a working hylafax/iaxmodem/zaphfc configuration ? Thanks in advance. ___ --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] 0.2.0-RC8o (* 1.0.9) + No Caller ID
Giovanni Miano wrote: I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o and * 1.0.9 I dont recive callerid from TDM400P fxo port but isdn hasnt problems If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok is it bug of bristuff ? Maybe, why not try bristuff 0.2.0-RC8p ? For me works fine (tdm400p cid detection). maxx ___ --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] Asterisk on AMD64
Joseph ha scritto: On Mon, 2005-09-12 at 09:56 +0200, Domjan Attila wrote: On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote: On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote: Does anybody runs Asterisk on AMD64? I can compile it on Gentoo, and start Asterisk a command line but as soon as I connect any device (like Sipura ATA ), asterisk crashes. Runs well here. here too :) What version are you running? I think you can't run asterisk on a 64 bit linux version. Certainly it works well on a 64 bit amd processor, but in x86 mode... Am i right guys ?? ___ --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: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?
Alessio Focardi ha scritto: Hello Lars, Have you got kernel sources installed ? I think that are mandatory for Zaphfc. Regards Not only, you have to have the kernel config save file too. Remember to make dep too. ___ --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: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?
Lars Dybdahl wrote: I would like to know how to install asterisk 1.0.9 with zaphfc working on a SuSE 9.2. Any ideas? Forget RPM. First of all read: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+SuSE then download http://www.junghanns.net/downloads/bristuff-0.2.0-RC8n.tar.gz explode the tarball, read the file INSTALL and run install.sh I suggest you to start and stop asterisk using the following script: http://www.leals.com/~mm/asterisk/asterisk_suse.sh.. Pay attention for loading zaptel modules correctly. maxx ___ Asterisk-Users mailing list Asterisk-Users@lists.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-cm-0.5 release announcement
Yes, I know. I was meaning the software thing. Diva server cancels echo via dsp only with new revisions boards (older boards are not able to run newer drivers with echo cancellation). Fritz cards don't cancel echo anyway. And echo squelch is only a trick that doesn't really solve the problem. Is it possible to port zap echo cancelor to different channels like chan_capi ? Armin Schindler ha scritto: On Thu, 23 Jun 2005, Massimo De Nadal wrote: Have you planned to integrate some echo cancel feature ? Echo cancelling (if the card supports it) is already implemented. As far as I know the Eicon Diva Server cards are the only cards supporting echo cancel via onboard DSPs. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.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-cm-0.5 release announcement
Sergio Chersovani wrote: As far as I know the Eicon Diva Server cards are the only cards supporting echo cancel via onboard DSPs. AVM active cards do not support it? No. Avm active cards are basically multi fritz boards running the same firmware onboard instead of charging pc cpu. They are surely more stable than fritz cards, but offer the same features (even with more channels). ___ Asterisk-Users mailing list Asterisk-Users@lists.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-cm-0.5 release announcement
Armin Schindler ha scritto: Which boards don't support that? If DSPs on board, echo-cancel should be available. I have in my hands right now a DIVA Server BRI-2M-PCI (not the 2.0 version) which own its dsp but doesn't echo cancel, due to old capi drivers which don't support this feature. Newer eicon drivers won't run on this board. Yes, that should be possible. But I don't think a channel driver (and each channel driver) should do that on its own. Software echo cancelling belongs in a common part of Asterisk. I strongly agree. But asterisk doesn't seem to work this way. Zap channel has it's own echo cancel engine. Other channels don't. This is so sad :-( Why not implement a really common echo cancel api usable from any channel ?? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] *67 with Sipura 3000
Change the dialplan in your spa3k with something like:
(xx.|*x.|**x.)
This way you can dial any number, even starting with * or **
Martin Roy ha scritto:
How can I dial *67 on a Sipura 3000 if I dial from a SIP phone
connected on an asterisk server. I always get a message saying that
authentication failed for INVITE for [EMAIL PROTECTED] If I dial a
number without doing *67 it's working fine...
sip 221 being the extension of my Cisco phone and 192.168.1.6 being
the IP of my asterisk server...
I have my outgoing context configure like this :
[outgoing]
ignorepat = 9
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,5,Playback(invalid)
exten = _9.,6,Hangup
When I do 9*67 and the number it take a while and then it will play
the invalid sound file and then hangup.
I even tried adding a second outgoing with this but it doesn't make a
difference :
exten = _9*67.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9*67.,5,Playback(invalid)
exten = _9*67.,6,Hangup
I figured that's it is a function of the Sipura 3000 but can I
disable it and make it seen as a number? Since Bell understand *67 I
don't need the Sipura 3000 to do something special with it...
Thanks
Martin
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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
Have you planned to integrate some echo cancel feature ? Armin Schindler ha scritto: Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax feature (provided by Frank Sautter) - auto-config (compile and work with Asterisk CVS-HEAD and older versions) - use with ELinOS cross-toolbox and project handling For the versioning, I have decided to use the name extention 'cm' to avoid confusion with kapejod's version. This first release is 0.5 (not 0.1) because the base is 0.4.0. Only the major and the minor number will be used. The exception to have a third number (patch-version) will be added for fixup-patches only. Feedback welcome. Armin PS: sorry for cross-posting. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early B3 connects on zaphfc
every day I discover something new about chan_capi and zaphfc. That's really exiting but, is there a place with some docs about these fine pieces of sw without asking for detail every time ?? Klaus-Peter Junghanns wrote: Hi, zapata.conf: prindication = passthrough best regards Klaus -- Klaus-Peter Junghanns ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ISDNguard
Anybody knows what is and how to use the ISDNguard daemon included in new bristuff packages ? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Spa3000 doesn't hangup after a conversation
Hi, I'm using a SPA 3000 as FXO and FXS termination connected to my * box. I'm using the caller ID prefix trick explained here: http://www.voip-info.org/wiki-Sipura+3000. All seems to work really fine, there is only a problem when I hangup my IP phone after a conversation and the other party don't: in this case the PSTN line remains connected until the other party hangs up. Any ideas ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH patch for bristuffed *
Anybody knows how to patch the music on hold bug on a bristuffed-0.2.0-RC7j 1.0.6-asterisk ? Thanks maxx ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Callgroup with bristuff ISDN?
Remco Barende wrote: Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? Can I use callgroups in such a setup, any config examples? Which isdn channel are you using ? Chan Capi, Zaphfc, mIsdn, isdn4l ?? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Callgroup with bristuff ISDN?
Remco Barende ha scritto: Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? How will this work for ISDN BRI/PRI? I don't want some extensions to get all calls from the BRI/PRI, just the calls from one DID. The wiki gives an example whereby a callgroup= is linked to a channel but this seems kinda silly with ISDN. Can I use callgroups in such a setup, any config examples? Thanks!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Sipua SPA-2000 and liong delay after dialling number
Simply dial a # after the number. Remco Barende ha scritto: When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.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] two avm usb isdn fritz v2.0 cards
Milos Kocbek wrote: I have a problem trying to install two avm fritz cards on one asterisk machine. I am using fcusb2 driver. 1 card works perfectly. The multiple fritz hack works only with pci cards (and, of course, it's a hack). Avm decided to not allow multiple installation for fritz cards, so I think it's not very easy to make 2 usb fritz work together. Regards maxx ___ Asterisk-Users mailing list Asterisk-Users@lists.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] bri stuff and unknown signalling type
Marco Parmeggiani wrote: I've downloaded and compiled zaphfc and libpri. To do that i've downloaded bri-stuff and commented out the asterisk related stuff because i've installed it from a debian package. Does this means that i have to rebuild the whole asterisk thing to support zaphfc? thanks ciao Zaphfc is a patch for asterisk. You don't only need to recompile the whole stuff, you have to be sure that you have the right asterisk version (depending on zaphfc version). Regards maxx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with a new italian service provider...
I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make asterisk working properly due to some asterisk authentication features lacks. The problem persists using chan_sip2 too. So, the question is: how can I help investigating the problem to obtain some good bugs report for digium ?? maxx ___ 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] mISDN, CAPI, ISDN ???
Definitely choose chan_capi. Chan_modem is almost deprecated, bad quality and very few features. Chan_misdn seems to be a very good project but it is still young. Zaphfc in theory it's wonderful (zap echo cancellation, timing etc.) but you have to use older * versions, (till new kapejod's release) and here in Italy (with italian nt1) I have many stability issues. Chan_capi works really great, you have to choose isdn boards with good capi drivers (avm, eicon) but the results is really stable and full featured. regards maxx Erwan DESVERGNES wrote: Did someone have experience with: - Chan_modem - Chan_capi - Chan_misdn What is the best??? **_** **Erwan Desvergnes **- **ANDIUM **- //82/86 rue Château Gaillard// //69100 Villeurbanne// //Tel. 04 37 43 44 45 / Fax 04 37 43 44 44// E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mISDN, CAPI, ISDN ???
Dave Cotton wrote: On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote: I've just used chan_capi it's very easy to use with Fritz!Cards and therefore I like it ;-) Worked straight out of the box on an AVM C2, hope it does the same with 2 Fritz!Cards in the same machine. Sadly no. If you want to use 2 fritz! in the same box you have to do a little hack with the drivers. http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO maxx ___ 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] mISDN, CAPI, ISDN ???
only avm active cards permit multiple installation straight forward (b1, c2 and c4) with fritz! pci you can do the hack mentioned above, for fritz! usb you can't install more then one. This limitiation is due to avm drivers design, they choose to allow multiple installation only on hi-end boards. Erwan DESVERGNES wrote: I don't know for the C2 but for the USB one it doesn't. AVM says it's normal. -Message d'origine- De : Dave Cotton [mailto:[EMAIL PROTECTED] Envoyé : mardi 19 octobre 2004 15:53 À : Asterisk List Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ??? On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote: I've just used chan_capi it's very easy to use with Fritz!Cards and therefore I like it ;-) Worked straight out of the box on an AVM C2, hope it does the same with 2 Fritz!Cards in the same machine. ___ 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] mISDN, CAPI, ISDN ???
yes, It's not the same, but applying the same hack to newer drivers it's not so difficult, almost for pci fritz! Erwan Desvergnes wrote: Seem It doesn't work for the USB one. And for the pci one, the current drivers it's not then same. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Massimo De Nadal Envoyé : mardi 19 octobre 2004 16:24 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ??? Sadly no. If you want to use 2 fritz! in the same box you have to do a little hack with the drivers. http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO maxx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Try deleting the line
pritrustusercid=yes
in zapata.conf
maxx
- Original Message -
From: Bastian Schern [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 8:34 PM
Subject: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Hi there,
I've got a small problem with the zaphfc channel. No MSN of an any
incoming call which comes trough the ISDN card (Acer ISDN, with HFC
chipset and zaphfc driver) which will be forwarded to the SIP-Phone will
be displayed. Always it will be shown asterisk an the Display.
--- snip (zapata.conf) ---
[channels]
language=de
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
pritrustusercid = yes
echocancel=yes
immediate=no
group = 1
context=default
channel = 1-2
--- snap ---
--- snip (extensions.conf) ---
[general]
static=yes
writeprotect=yes
[globals]
BASTIAN=SIP/16
[macro-callwithmsn]
exten = s,1,SetCallerID(${ARG2})
exten = s,2,SetCIDName(${ARG3})
exten = s,3,Dial(Zap/g1/${ARG1},60,Ttr)
exten = s,104,Playtones(busy);
exten = s,105,Busy
[default]
exten = 96,1,SetCIDNum(${CALLERIDNUM})
exten = 96,2,Dial(SIP/16)
exten = _0.,1,Macro(callwithmsn,${EXTEN:1},61,Bastian)
exten = _XX,1,Dial(SIP/${EXTEN})
--- snap ---
It would be very nice if somebody can help me.
Regards
Bastian
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Re: [Asterisk-Users] GrandStream BT101 Attended Transfers
I've asked Grandstream tech support about attended transfer. They told me that in about a month there will be available a firmware upgrade that supports attended transfer natively. maxx Chris Shaw wrote: I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling zaphfc
You cannot compile zaphfc with latest CVS head. You have to donwload specific date version using the download.sh included script. BTW I have some problems with RC4. It works fine with my 2 isdn pci boards, but it seems to be unable to drive my TDM400 ... Try RC3, at the moment seems to be more stable. maxx Christian Victor wrote: Hi! I have a problem compiling the zaphfc driver for my HFC-PCI cards. I use Asterisks latest CVS and bri-stuff.0.1.0-RC4. The install.sh compiles zaptel and libpri without problems. But when it tries to compile qozap and zaphfc it show the following errors: qozap.c:206: error: structure has no member named `bytes2transmit´ qozap.c:211: error: structure has no member named `eoftx´ the error is repeated in a few lines. Then qozap.c:374: error: structure has no member named `bytes2receive´ qozap.c:377: error: structure has no member named `eofrx´ wich is also in multiple lines of qozaop.c then ´qotap.c:617: error: `ZT_FLAG_BRIDCHAN´ undeclared (first use in this function) The same happened with tha letest stable release of Asterisk and with bri-stuff RC2k and RC3. Could anybody help please? Thanks a lot Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling zaphfc
forget the asterisk source you have downloaded. Zaphfc is not only a driver, it's a patch that have to be applied to specific source version too, You have to run the install.sh script that is included in the tarball. This script before downloads the right asterisk version (download.sh) and then compiles it (compile.sh). BTW RC4 is buggy, download RC4a that works fine to me. maxx - Original Message - From: Christian Victor [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 5:03 PM Subject: Re: [Asterisk-Users] Problem compiling zaphfc Just a little update. I installed Asterisk stable of 08/19/04 and tried to compile bri-stuff RC3 and RC4 with it. Same problem as described below. I cant't believe that my whole asterisk setup is riuned by that §$%%$ ISDN drivers. ;-) Maybe the cause for the problem is a missing library or something. But I think I installed everything that is known to be required under Debian Sarge. Maybe someone of you has a clue. Chris Christian Victor schrieb: You cannot compile zaphfc with latest CVS head. You have to donwload specific date version using the download.sh included script. BTW I have some problems with RC4. It works fine with my 2 isdn pci boards, but it seems to be unable to drive my TDM400 ... Try RC3, at the moment seems to be more stable. But how can I now wich version of Asterisk I have to install? For what date version is bri-stuff.RC3 made? Chris I have a problem compiling the zaphfc driver for my HFC-PCI cards. I use Asterisks latest CVS and bri-stuff.0.1.0-RC4. The install.sh compiles zaptel and libpri without problems. But when it tries to compile qozap and zaphfc it show the following errors: qozap.c:206: error: structure has no member named `bytes2transmit´ qozap.c:211: error: structure has no member named `eoftx´ the error is repeated in a few lines. Then qozap.c:374: error: structure has no member named `bytes2receive´ qozap.c:377: error: structure has no member named `eofrx´ wich is also in multiple lines of qozaop.c then ´qotap.c:617: error: `ZT_FLAG_BRIDCHAN´ undeclared (first use in this function) The same happened with tha letest stable release of Asterisk and with bri-stuff RC2k and RC3. Could anybody help please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaphfc CallerID problem...
Thorsten Huber wrote:
Hi,
On Sun, Aug 01, 2004 at 09:13:52PM +0200, Massimo De Nadal wrote:
...
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,3,HangUp
The problem is that when I receive a call, I can't see the CallerID
neither
on the Cisco 7940 nor on the X-Lite client.
Any ideas ??
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,4,HangUp
--
Gruss / Best regards | LF.net GmbH| fon +49 711 90074-414
Thorsten Huber| Ruppmannstrasse 27 | fax +49 711 90074-33
[EMAIL PROTECTED] | D-70565 Stuttgart | http://www.lf.net
Thank you for your answer. I think my problem is a little bit different.
If I add a priority like:
exten = s,2,NoOp (${CALLERID}_${CALLERIDNUM}${CALLERIDNAME})
I can see that CALLERID and the other variables have no value it seems
like channel zap knows the CALLERID but doesn't set the variables values.
Maybe a zaphfc bug ?
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Re: [Asterisk-Users] Zaphfc CallerID problem...
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,4,HangUp
Thank you for your answer. I think my problem is a little bit different.
If I add a priority like:
exten = s,2,NoOp (${CALLERID}_${CALLERIDNUM}${CALLERIDNAME})
I can see that CALLERID and the other variables have no value it
seems
like channel zap knows the CALLERID but doesn't set the variables
values.
Its the same on our installation. Instead of zaphfc we're using quozap
with a QuadBRI. And its working with SetCIDName.
Maybe a zaphfc bug ?
It could also be a wanted behavior. There's antother Question: Do you
always trust the received number (this applies to all channel types)?
Thanks for your help again.
Well, I'll try to explain myself better (I'm really sorry for my bad english
:-)
If I try to: exten =s,2,SetCIDName(FOOBAR) , when I receive a call on phone
display I can read FOOBAR correctly, but the variable CALLERID (which is
expanded by the syntax ${CALLERID} ) contains no values at all
Yes, I always want to trust received number. What I need is to read on the
phone the caller number. If the number is masqueraded or hacked or other, it
doesn't matter.
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Re: [Asterisk-Users] Zaphfc CallerID problem...
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,4,HangUp
Thank you for your answer. I think my problem is a little bit
different.
If I add a priority like:
exten = s,2,NoOp (${CALLERID}_${CALLERIDNUM}${CALLERIDNAME})
I can see that CALLERID and the other variables have no value it
seems
like channel zap knows the CALLERID but doesn't set the variables
values.
Its the same on our installation. Instead of zaphfc we're using quozap
with a QuadBRI. And its working with SetCIDName.
Maybe a zaphfc bug ?
It could also be a wanted behavior. There's antother Question: Do you
always trust the received number (this applies to all channel types)?
Thanks for your help again.
Well, I'll try to explain myself better (I'm really sorry for my bad
english
:-)
If I try to: exten =s,2,SetCIDName(FOOBAR) , when I receive a call on
phone
display I can read FOOBAR correctly, but the variable CALLERID (which is
expanded by the syntax ${CALLERID} ) contains no values at all
Yes, I always want to trust received number. What I need is to read on the
phone the caller number. If the number is masqueraded or hacked or other,
it
doesn't matter.
OK!!! I've solved the problem removing PriTrustUserCID=yes from zapata.conf
.
Now CID works great !
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[Asterisk-Users] Zaphfc CallerID problem...
I'm not sure that this problem is strictly related to zaphfc, but this is what happens: my asterisk (build on bri-stuff-0.1.0-RC2k) handles a single PCI HFC-S based card. I own a Cisco 7940 Sip phone (fw 7.1) and a pc running X-Lite. Zaptel.conf and zapata.conf are taken directly from zaphfc samples. Extension.conf contains the following lines: [from-ISDN1] exten=s,1,Wait(1) exten=s,2,Dial(Sip/cisco1Sip/xlite1,30,tTr) exten=s,3,HangUp The problem is that when I receive a call, I can't see the CallerID neither on the Cisco 7940 nor on the X-Lite client. Exactly the cisco phone tells me: From asterisk / asterisk x-lite: Call incoming on line asterisk The strange think is that on asterisk console when a call arrive I see: - Accepting call from 'xx' to 's' on channel 0/1, span 1 - Executing Dial (Zap/1-1,Sip/cisco1Sip/xlite1|30|tTr) (where is the correct caller id) Any ideas ?? ___ 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] Daytime - Nighttime
Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ??? ___ 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] Daytime - Nighttime
Ok guys... the example is simply perfect ! Thanks a lot and shame on me for not reading carefully the wiki :-) maxx On Thu, 22 Jul 2004 14:54:29 +0100, Steve Hanselman [EMAIL PROTECTED] wrote: Yes, you'd have a dialplan entry that set a value in the database, then acted upon that. You'd probably want some nice voice prompts The system is currently in [Day/Night/Holiday] mode, press 1 to set to day, 2 to set. Here is the start of a simple one I'm sure you will be able to extend it from this http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant ___ 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] Bri solution for Asterisk
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? Have anyone tried chan_mISDN on a 2.6 box ? How does it run ??? ___ 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] Bri solution for Asterisk
going to i4l means... incoming sound sometimes gets interpreted as DTMF - and when your caller humms a '#' - transfer kicks in... Outgoing DTMF mhhh almost unuseful but surely funny ;-) There is an Update patch for bristuff... look carefully in the download directory. do you mean bri-stuff-0.1.0-RC1 ?? I've tried out this release, but it seems to be bugged. After 8-10 seconds of correct work I get the message Primary D-Channel on span 1 down and the isdn card stops to work. How can I tell kapejod about the bug ? ___ 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] Problem with asterisk and zaphfc
Hi everybody, I have a problem using zaphfc. When I start asterisk after 8-10 seconds I get the message Primary D-Channel on span 1 down and my isdn modem stops to work. If I place or receive a call before this message all works really fine (even if the call is very long), but when I hang up, after a few seconds I get the message and the modem crashes. Stopping and restarting asterisk solves the problem for another 8-10 seconds :-( Thank you for any suggestions maxx ___ 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
