Re: [asterisk-users] Please suggest me Best VoIP Service Provider
On Jul 18, 2006, at 10:58 PM, Crazy Boy wrote:Hi Friends,I am Chandra from India. Currently, we are implementing Asterisk in our organization and using "Teliax" service to make calls to USA. Initially, Teliax service is very good. But, from 20 days, Teliax service is very poor and they are not responding for customer querys also. So, we want to change the VoIP Service provider. Our main aim is to make calls to USA from INDIA using "Asterisk". Can you please suggest me some best VoIP Service providers? Looking forward to your response. Thank you.I also use teliax, and have found them to be quite decent in the service and support department and pretty reliable as a provider.That said, I also discovered that I had some call quality issues, which where traceable to the route between by asterisk box and them (teliax).It's very important to carefully analyze your routes and make sure that whatever provider/terminator you are using has as short and clean a route as possible.I don't know how well that will work from India to the US?I found a terminator called sellvoip.net, whose website is crap(currently), but whose route from my server is very clean and short.My calls all sound perfect now. I keep teliax and nufone configured as backups, and they both largely work well, but not as well as my shortest route.Short version, there is no such thing as a "good terminator" without looking at your route to them.Good Luck,Marty___ --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] Keep Zap Channel from answering
Anyone know how to keep an Analog Zap channel from answering? I know I can answer it and send it to voicemail or do any number of other things with it once it's answered. I want to keep Asterisk from answering it, completely ignoring it while still having the line connected for outgoing purposes. Reason is, I have Vonage line I am going to be porting and for now it works horribly for inbound calls hooked up from the Cisco 186 - Wildcard. What I have done is setup an instant forward with Vonage to another number with another provider. Problem is, Vonage still rings the ATA once causing the call to be picked up by Asterisk instead of being forwarded as intended. I know, I could just unplug the ATA but it's bugging me and I would like to use it for outbound until I port the number and close the account. Looked around quite a bit but I can't find much on this topic. ___ --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] Please suggest me Best VoIP Service Provider
Hello chandra, What is your volume and target, we could provide you with USA route using your asteriks gbenga ---Original Message--- From: Crazy Boy Date: 07/19/06 06:59:19 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Please suggest me Best VoIP Service Provider Hi Friends,I am Chandra from India. Currently, we are implementing Asterisk in our organization and using "Teliax" service to make calls to USA. Initially, Teliax service is very good. But, from 20 days, Teliax service is very poor and they are not responding for customer querys also. So, we want to change the VoIP Service provider. Our main aim is to make calls to USA from INDIA using "Asterisk". Can you please suggest me some best VoIP Service providers? Looking forward to your response. Thank you.Regards,Chandra. Groups are talking. Were listening. Check out the handy changes to Yahoo! Groups. ___ --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] Please suggest me Best VoIP Service Provider
Take this to the -biz list, PLEASE. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Uni Call
Hi Guy Does anyone know where I can find a patch for the latest unicall and asterisk 1.2.7.1 Regards Mark Brooker ___ --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 + fax
If there any way to pass on that problem, like i know the source should cancel the echo on the line. In addition i am trying to connect regular fax through ata to asterisk with no success. Regular Fax machine - ata - Asterisk. ata is registering to the asterisk as regular extension. Instead of phone after the ata i have a fax machine. i am trying to send a fax to my ATA FAX MACHINE with no success it's falling after the dialing i don't see the connection stage. I think i am missing something. Any help will be appreciated. The ATA must be T.38 enabled. For instance: PAP2 is not. Benchev ___ --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] Ztdummy
Using Zaptel-1.2.7 Asterisk 1.2.10 OS: CentOS 3.4 I am having a problem trying to get ztdummy and it wont work. Here is what I did the following and got: [EMAIL PROTECTED] ~]# modprobe zaptel[EMAIL PROTECTED] ~]# modprobe ztdummyNotice: Configuration file is /etc/zaptel.confline 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy[EMAIL PROTECTED] ~]# What I find interesting is that timing will work. However I don't feel comfortable letting the client use the system if this can affect him in anyway. Thanks. Dovid ___ --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] header replacement
Hi, I know there are commands called sipaddheader and sipgetheader but I can't find any method to replace/remove header. Does anyone know how to replace/remove a header in the sip message? Say, I have a header field in the sip message: Remote-Party-ID: [EMAIL PROTECTED] I want to replace it to :Remote-Party-ID: [EMAIL PROTECTED] Thanks ___ --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] Zaptel Compilation Error
Hi to all, Has anybody in this group encounter this kind of error in their recompilation of zaptel. My asterisk box has a for E1/T1 digium card with echo cancellation. I want to upgrade the zaptel driver. So I downloaded the new driver from digium and recompile it but I got errors. Please the errors below: ZAPTELVERSION=1.2.7 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp /lib/modules/2.6.9-34.0.2.ELsmp/build make -C /lib/modules/2.6.9-34.0.2.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1034: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1037: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1047: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1054: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1107: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1188: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1211: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1584: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1620: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:3343: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:3345: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_init': /usr/src/zaptel/zaptel.c:6553: error: incompatible types in assignment /usr/src/zaptel/zaptel.c: At top level: /usr/src/zaptel/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-i686' make: *** [linux26] Error 2 Did I miss something? Please help. Regards, Leonimar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Zaptel Compilation Error
Think this has been covered several times on the list. Sounds like the spinlock.h issue. You need to go into the kernel directory, for you it seems like the /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/linux/spinlock.h file and replace anywhere you see 'rw_lock' with 'rwlock', make clean, make On 7/19/06, leonimar cape [EMAIL PROTECTED] wrote: Hi to all, Has anybody in this group encounter this kind of error in their recompilation of zaptel. My asterisk box has a for E1/T1 digium card with echo cancellation. I want to upgrade the zaptel driver. So I downloaded the new driver from digium and recompile it but I got errors. Please the errors below: ZAPTELVERSION=1.2.7 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp /lib/modules/2.6.9-34.0.2.ELsmp/build make -C /lib/modules/2.6.9-34.0.2.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1034: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1037: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1047: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1054: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1107: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1188: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1211: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1584: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1620: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:3343: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:3345: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_init': /usr/src/zaptel/zaptel.c:6553: error: incompatible types in assignment /usr/src/zaptel/zaptel.c: At top level: /usr/src/zaptel/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-i686' make: *** [linux26] Error 2 Did I miss something? Please help. Regards, Leonimar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
[asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
I've been trying to install bristuff on my system for a really long time. This is what I've done so far: I started with a [EMAIL PROTECTED] installation. I tried downloading and compiling bristuff release - it didn't work. It was a long time ago, I don't remamber what the problem was. I tried compiling the latest bristuff (whatever latest was about 1-2 months ago). It failed to compile. I download the full CentOS 4.3 and tried compiling both bristuff release (0.2.0-RC8) and bristuff latest. Again, whatever latest was about 2 weeks ago. Next I found something about bristuff being known to work on kernel 2.4; Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried compiling both bristuff release (0.2.0-RC8) and the current release of today (19 july 2006). I wasn't able to compile ither one of them. Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) and two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling everything a few times, I think it's time to ask for help: Would someone be so kind and tell my how they installed Bristuff from A to Z? (that is, what version of Linux so I can download the same version, what updates, what version of bristuff). I'm hoping for a quick answer like: Install LinuxVariant 10.20, install all updates using LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call install.sh and be done with it. Thanks, Cosmin Prund ___ --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] Alternative (?) ways to handle G.729 and annexb
Hello everybody, naive Asterisk user here, so please excuse my vast ignorance on the subject that follows. I would be more than happy to be corrected here, so implicitly an AFAIK is present in all of my sentences. :) As you (may already) know and AFAIK, G.729-enabled Asterisk responds to G.729 offers as follows: m=audio RTP/AVP X Y 18 Z ... a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no ... Notice that the response specifies essentially two things: 1) We concur to using codec 18 (G.729) and 2) We state we will not send and we are not willing to receive SID frames In a RFC context, the SDP answer above is correct, if the offer was something like this: m=audio RTP/AVP 18 K L Z ... a=rtpmap:18 G729/8000 a=mtp:18 annexb=no ... The problem here is that the other side might have sent an offer that implicitly (no reference to annexb=no) or explicitly (direct reference to annexb=yes) indicated Annex B behaviour. All the following are semantically equivalent according to Table 1 and Section 4.5.6 of STD0065 (RFC3551) and Section 4.1.9 of RFC3555: m=audio RTP/AVP X Y 18 Z ... no reference to payload 18 ... -OR- m=audio RTP/AVP X Y 18 Z ... a=rtpmap:18 G729/8000 No reference to fmtp:18 annexb=yes ... -OR- m=audio RTP/AVP X Y 18 Z ... a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes ... In all the examples above, sender requests Annex B behaviour and will most likely send SID frames. Asterisk accepts the comms, but will drop (perhaps) SID frames. End result: worse voice quality. Thoughts so far: 1) Perhaps I'm totally wrong here, but shouldn't 18 in these last cases be dropped, since annexb behaviour is not supported by us - an important characteristic of the codec offered is not supported by us, hence this is not an attribute that could be changed in asterisk's answer - 18 should be dropped altogether in our response? 2) Quite a few clients out there do not send a=fmtp:18 annexb=no (a=fmtp:XX annexb=no), meaning they are Annex B capable, whereas they are not... This must be taken into account (current Asterisk implementation does not present any problems here, non-Annex-B comms will be established both ways) since in the future Asterisk might support Annex B as well... With these in mind, some criteria for a robust (it's a joke actually, since my knowledge is vry limited, so like I said please spare me :) ) G.729 implementation would be: 1) Best sound quality possible 2) Recognition of UAs at fault which are not sending annexb info Possible pcode for an Asterisk implementation with no Asterisk AnnexB functionality: for all codetypes offered by UA in initial INVITE if codetype is G729 then if exists(annexb) and annexb=no then proceed with normal SDP parsing else drop codenumber which correspond to offered G729 if this is the only codenumber offered then respond with error code and terminate call exit endif // Note that AFAIK the UA can specify multiple codetypes for G729 // one for Annex B with annexb=yes, another for non-Annex-B, // with annexb=no select an unused (from both the offer and the answer) codenumber above 96 say ZZ // to isolate the case where the UA does not send annexb, even though // it is not annexb compliant, we are offering a new codepoint which in // in the answer's m SDP line would have the same priority as if // we accepted G.729 in the first place include in the response: a=rtpmap:ZZ G729/8000 a=fmtp:ZZ annexb=no endif endif endif end for rest of code send RE-invite to UA, specifying only: a=rtpmap:ZZ G729/8000 a=fmtp:ZZ annexb=no Sorry for not making sense, my English could be better. Any opinions, especially from developers would be kindly appreciated. Regards, M.- ___ --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] Keep Zap Channel from answering
voiplist wrote: Anyone know how to keep an Analog Zap channel from answering? I know I can answer it and send it to voicemail or do any number of other things with it once it's answered. I want to keep Asterisk from answering it, completely ignoring it while still having the line connected for outgoing purposes. assuming the line is attached to the trunk context, try the following in your dialplan: [trunk] exten = s,1,Congestion exten = s,2,Hangup Flynn ___ --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] header replacement
Hi, I know there are commands called sipaddheader and sipgetheader but I can't find any method to replace/remove header. Does anyone know how to replace/remove a header in the sip message? Say, I have a header field in the sip message: Remote-Party-ID: [EMAIL PROTECTED] I want to replace it to :Remote-Party-ID: [EMAIL PROTECTED] Thanks ___ --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] Unicall libmfcr
Hi Im wondering if you can help me with this error mfcr2.c:3543: error: `UC_REQUESTMOREINFO_ORIGINATING_NUMBER' undeclared (first use in this function) mfcr2.c: In function `call_control': mfcr2.c:3894: error: `UC_OP_REQUESTMOREINFO' undeclared (first use in this function) mfcr2.c:3895: error: `uc_requestmoreinfo_t' undeclared (first use in this function) mfcr2.c:3895: error: syntax error before ')' token make[1]: *** [mfcr2.lo] Error 1 make[1]: Leaving directory `/usr/src/unicall/libmfcr2-0.0.3' make: *** [all] Error 2 Ive got the latest snapshot 20060205. Regards Mark Brooker ___ --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] Please suggest me Best VoIP Service Provider
Martin Joseph wrote: I found a terminator called sellvoip.net, whose website is crap(currently), but whose route from my server is very clean and short. My calls all sound perfect now. I keep teliax and nufone configured as backups, and they both largely work well, but not as well as my shortest route. What codec are you using with sellvoip? I have to use G729 but I find that while the calls are setup, I get one-way audio on every call. The called party cannot hear me. Let me know your config if you would. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] attended transfer issue
Thomas Artner wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
I've just come across this issue too. As the call gets hung up if the
transfer is attempted before answer I tried changing this:
exten = _90ZX,1,Dial(zap/g1/${EXTEN},,TW)
to this:
exten = _90ZX,1,Answer
exten = _90ZX,n,Dial(zap/g1/${EXTEN},,TW)
So the call is 'answered' in one sense before it starts ringing. I've
only tested it on a zap channel so far but it seems to fix it. Unless
this is how Answer() is supposed to be used, I'm not sure then it's a
bit of a dirty hack and I don't know what else it might break.
I'm not back in the office until next week so can't test my brainwave
out fully.
Mike
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[asterisk-users] Issue with g729 codec
Hi All, I have a problem with conferencing, but it's more to do with the g729 codec. I have purchased six licenses for g729 for all our phones, and occasionally want to do conferencing, but at the moment it only allows two people in before the licenses run out. When two people are in the conference and I do a 'show g729' I get the following: *CLI show g729 2/6 encoders/decoders of 6 licensed channels are currently in use And when another person joins the conference they can listen but are unable to speak because all 6 decoders licenses are used up. Any ideas at all from anyone how to fix?? It occurs with all the version I've tried from 1.0.7 to 1.2.1 1.2.7 and 1.2.10. Regards, Daniel ___ --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] QuadBRI + TDM + GSM hangup problems
Hello everybody, I have an Asterisk (1.2.9.1-BRIstuffed-0.3.0-PRE-1q) with a QuadBRI, a TDM400P with 2 FXO modules, and 2 Nokia 22 GSM modules. The system can receive and make calls perfecty, via ISDN and GSM. But when I configure Asterisk to redirect the calls from ISDN to a mobile telephone (via the GSM modules), when the mobile phone answers, the call is terminated: --- Zap/7 is the ISDN (g0 in zapata.conf) Zap/15 is the GSM (g1 in zapata.conf) [...] Jul 19 11:18:54 VERBOSE[28704] logger.c: -- Called g1/660XXX Jul 19 11:18:55 DEBUG[28704] chan_zap.c: Exception on 14, channel 15 Jul 19 11:18:55 DEBUG[28704] chan_zap.c: Got event Hook Transition Complete(12) on channel 15 (index 0) Jul 19 11:18:56 DEBUG[28704] chan_zap.c: Exception on 14, channel 15 Jul 19 11:18:56 DEBUG[28704] chan_zap.c: Got event Dial Complete(9) on channel 15 (index 0) Jul 19 11:18:56 DEBUG[28704] chan_zap.c: Enabled echo cancellation on channel 15 Jul 19 11:18:56 DEBUG[28704] chan_zap.c: Engaged echo training on channel 15 Jul 19 11:18:59 DEBUG[28704] chan_zap.c: Exception on 14, channel 15 Jul 19 11:18:59 DEBUG[28704] chan_zap.c: Got event Dial Complete(9) on channel 15 (index 0) Jul 19 11:18:59 DEBUG[28704] chan_zap.c: Echo cancellation already on here is the moment the mobile phone answers, the ISDN channel hangup Jul 19 11:19:18 VERBOSE[27851] logger.c: -- Channel 0/1, span 3 got hangup, cause 31 Jul 19 11:19:18 DEBUG[28699] app_dial.c: Unable to forward frame Jul 19 11:19:18 DEBUG[28699] app_dial.c: Exiting with DIALSTATUS=CANCEL. Jul 19 11:19:18 DEBUG[28704] chan_zap.c: Hangup: channel: 15 index = 0,normal = 14, callwait = -1, thirdcall = -1 Jul 19 11:19:18 DEBUG[28704] chan_zap.c: disabled echo cancellation on channel 15 Jul 19 11:19:18 DEBUG[28704] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/15-1 Jul 19 11:19:18 DEBUG[28704] chan_zap.c: Updated conferencing on 15, with 0 conference users Jul 19 11:19:18 VERBOSE[28704] logger.c: -- Hungup 'Zap/15-1' Jul 19 11:19:18 DEBUG[28704] app_dial.c: Exiting with DIALSTATUS=CANCEL. Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/7-1 Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Hangup: channel: 7 index = 0, normal = 21, callwait = -1, thirdcall = -1 Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Already hungup... Calling hangup once, and clearing call Jul 19 11:19:18 DEBUG[28699] chan_zap.c: disabled echo cancellation on channel 7 Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/7-1 Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Updated conferencing on 7, with 0 conference users Jul 19 11:19:18 DEBUG[28699] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/7-1 Jul 19 11:19:18 DEBUG[28699] chan_zap.c: disabled echo cancellation on channel 7 Jul 19 11:19:18 VERBOSE[28699] logger.c: -- Hungup 'Zap/7-1' --- The zapata.conf is this: --- [channels] language=es context=default usecallerid=yes cancallforward=yes immediate=no musiconhold=default faxdetect=incoming useincomingcalleridonzaptransfer=yes ; TDM with 2 FXO modules signalling=fxs_ks busydetect=yes busycount=7 language=es context=from-pstn callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 callerid=asreceived callgroup=1 pickupgroup=1 group=1 channel=15-16 ; quadbri signalling=bri_cpe_ptmp context=from-pstn switchtype=euroisdn language=es pridialplan=local prilocaldialplan=local rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes faxdetect=incoming group=0 callgroup=1 pickupgroup=1 answeronpolarityswitch=yes hanguponpolarityswitch=yes immediate=no channel=1-2,4-5,7-8,10-11 -- Can someone explain me why the call is terminated? How can I solve this problem? Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --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] BudgeTone BT-102 not registering to Asterisk
Hello everyone, is there any known bug in the inner working of BT-10X series? It doesn't register to one of the testing Asterisk servers that I have at work, I don't understand why!! The logs simply say that it's sending SIP message 1 to the Asterisk Server: Jul 19 11:44:50 192.168.1.40 GS_LOG: [00:0B:82:09:5D:81][000][FFFB][01000817] Send SIP message: 1 To 192.168.1.2:5060 repeated N times (with N 10). The phone is configured correctly, with the username and password specified in sip.conf. Did someone encounter this issue? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Setting a threshold for asterisk to take ZAP line off hook ?
On 7/18/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I think when a PSTN line says 'Ring' it's simply for aesthetics... The line is 'answered' the instant * connects to it for two-way audio... (well not that instant but somewhere in the connection process. When you are hearing ringing from the PSTN through a zap card, the rings are coming from the phone company and are just sound. * doesn't decode that and act on it yet.) Maxim Vexler wrote: On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote: On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote: Hello list I'm trying to setup asterisk as an answering machine. How can I set asterisk to Answer() incoming call ONLY after specified count of ring cycles ? In the current situation I have the PBX connected to a home line, where POTS device are also connected on the same circuit. What I'm trying to do is allow a grace period where a POTS device could be picked up and those stop the ring indication on the line by this causing asterisk to not answer the call. In present situation even if the incoming phone call is taken off hook by a POST device asterisk still starts playing its incoming call IVR after the specified(where?) number of seconds. I don't think you can do that, since asterisk has no way to know when the shared PSTN line is answered by your analog phones... I don't think asterisk counts the rings, as much as it waits for answered status, which it is never going to see in your current configuration. I am a relative newb though, so maybe someone else here has a brilliant idea for you? ___ --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 You have a point but no way am I going to accept that as an answer. Here's the log off such case : Jul 16 21:59:20 DEBUG[4387] chan_zap.c: Monitor doohicky got event Ring Begin on channel 1 Jul 16 21:59:21 DEBUG[4387] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1 Jul 16 21:59:21 DEBUG[4387] dsp.c: dsp busy pattern set to 0,0 Jul 16 21:59:21 VERBOSE[4411] logger.c: -- Starting simple switch on 'Zap/1-1' Jul 16 21:59:21 DEBUG[4373] devicestate.c: Changing state for Zap/1 - state 2 (In use) Jul 16 21:59:21 DEBUG[4412] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Jul 16 21:59:29 WARNING[4411] chan_zap.c: CallerID returned with error on channel 'Zap/1-1' Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Answer' Jul 16 21:59:29 VERBOSE[4411] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Took Zap/1-1 off hook Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Enabled echo cancellation on channel 1 Jul 16 21:59:29 DEBUG[4411] chan_zap.c: No echo training requested Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Set' As you can see, the first two events are event Ring and event Ring/Answered. What I need is the driver of chan_zap.c counting 5 event Ring before starting Ring/Answered. It can't be that hard (I think). Thank you for your answer. -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 Well, I did managed to get it to work some how with the attached patch. My problems with this code are : a. It will destroy all active calls because I'm using zap_restart(). I really need to find a better way to destroy only the active channel instead of the whole zaptel stock. b. I don't know how this might be related, but since I've started to use this patch some calls simply won't get disconnected by asterisk after remote party hangup. Note that I'm using busydetect on this channel (It's x100p.com clone). Please note that when asterisk does not disconnect the call with busy detect, I'm seeing this in the full log (attached) the following : Jul 19 12:39:39 DEBUG[24397] channel.c: Scheduling timer at 160 sample intervals where as with normal calls this does not appear. This only occurs if I let asterisk Answer the call and instantly hangup my cell phone (which I'm using to test this). If on the other hand I listen to the IVR for a few seconds and then hangup my cell phone asterisk will take the channel On Hook using busydetect as expected. Meaning that I am able to reproduce this behaviour (bug?) by letting chan_zap take the call off hook followed by instant remote party (cell phone) hangup. diff -Naur asterisk-1.2.7.1.dfsg/channels/chan_zap.c asterisk-1.2.7.1.dfsg-anspatch/channels/chan_zap.c --- asterisk-1.2.7.1.dfsg/channels/chan_zap.c 2006-04-04 21:28:14.0 +0300 +++ asterisk-1.2.7.1.dfsg-anspatch/channels/chan_zap.c 2006-07-19
[asterisk-users] Dynamic Queue Members never called
Hi List,
I have a problem with dynamic queue members, they are never called when
a user is queued...
The queue works fine, with a cli command show queue testing I can see
that there's an user waiting
but there are no calls to members done...
Does anybody knows why or do I missed something ?
Thanks,
Tristan
--
Dialplan to enqueue user:
[test]
exten = s,1,Answer
exten = s,2,Dial(Local/[EMAIL PROTECTED]/n||g)
exten = s,3,NoOp(End QUEUE)
exten = enqueue,1,Answer
exten = enqueue,2,Queue(testing)
exten = h,1,NoOp(HANGUP IN TEST)
here is the way I configured my queue:
[testing]
strategy = leastrecent
timeout = 15
retry = 1
wrapuptime=0
context = listen-testing
eventwhencalled = yes
eventmemberstatus = yes
announce-frequency = 0
announce-holdtime = off
queue-callerannounce = queue-callerannounce
reportholdtime = no
servicelevel = 20
timeoutrestart = yes
--
I put on the cli:
CLI add queue member Local/[EMAIL PROTECTED] to testing
--
part of the dialplan:
[dial_queue]
exten = _,1,ChanIsAvail(Zap/R3)
exten = _,n,NoOp(AvailChannel=${AVAILCHAN})
exten = _,n,Set(__DialChannel=${CUT(AVAILCHAN,,1)})
exten = _,n,Dial(${DialChannel}/${EXTEN}|${DIALTIMEOUT}|M(agentanswer))
exten = _,n,GotoIf($[ ${DIALSTATUS} != ANSWER ]?stop|1)
exten = stop,1,Busy
exten = stop,2,NoOp(WARNING AGENT NO ANSWER)
[listen-testing]
exten = *,1,Hangup()
___
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[asterisk-users] Finding far end echo in Verizon network
This is a weird one. Network: Asterisk ver 1-0-9 on DL360. 10 Cisco 7960g phones with 3.8.2 SIP Load. Gateway - Cisco 2811 router with 4 x verizon bri's. Network - Private vlan with 1ms response times to all devices. Issue: Intermittent echo on outbound/inbound calls. Users hearing their own voice about 0.5sec later. Tried so far: Upgraded firmware on some phones to 3.8.2 Upgraded software on Cisco router. Changed gain and attentuation settings on cisco router Got Verizon to test bris Moved rtp from asterisk direct to phone and router (canreinvite=yes) load tested asterisk None of the above made any difference. They are hearing their own voice so that means the issue is on the far end. But should it be up to me to control the possible delay or slippage in the verizon bri network? Any help much appreciated. Taf. ___ All new Yahoo! Mail The new Interface is stunning in its simplicity and ease of use. - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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] Install H323
Hello, I advise you to install OH323 channel with gnugk . Have you got a h323 ip phone ? Harry --- Wasif [EMAIL PROTECTED] a écrit : Hello, I just downloaded Tribox 1.1 having Asterisk 1.2.9.1. I need to have H323 support with asterisk like sip. Please guide me how I can do this. Thanks Wazb ___ --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 ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --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] Issues with MeetMe
Hi, Having a few problems getting MeetMe conferencing to work. I'm able to get the conference established, get to the point where I can hear the message You are the only person in this conference.. Asterisk then segfaults. The only thing printed to the console during this time is: -- Executing MeetMe(SCCP/flat-1-0008, ) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '681' -- Playing 'conf-onlyperson' (language 'en') Segmentation fault [EMAIL PROTECTED]:~# Ouch ... error while writing audio data: : Broken pipe Any ideas? I'm running Asterisk v1.2.9.1, on Centos with a Linux 2.6.15 kernel. There's an X100P in the system, so I'd assume timing isn't an issue. Rgds, Chris Jones // Network Administrator Top Level Internet e: [EMAIL PROTECTED] ___ --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] Parked calls
Hello everybody, I is possible to manage multiple call parked per line . I mean a caller (agent) have to park more than two call . It is possible to retrieve caller one ,two ,three, ... with a aplliction which one display the calling parked to a PC screen or a screen phone . Regards Harry ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --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] Question setting up a bat phone extension.
Kai Ober wrote: Eric ManxPower Wieling schrieb: Grandstream seems unable to produce stable firmware. They have tried for *YEARS* and still people have to try many different versions of the firmware to find one that actually works in their environment. okay, i see, thx :) i will try to remember, if i'm ever going to buy an VoIP-Phone. any suggestions for this situation? (i.e. which devices do you prefer) Polycom, Cisco, SIPura/Linksys. I don't like Cisco's firmware licensing, but they are still good phones. Polycoms is the brand of phones we use, SIPura is the brand of ATAs we use. Many people like the Linksys/SIPura phones. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Re: CentOS 4.3 and Zaptel-1.2.7
In article [EMAIL PROTECTED], Russ Price [EMAIL PROTECTED] wrote: varun wrote: I have problems compiling zaptel 1.2.6 on my CentOS 4.3. CentOS is updated and I believe I have installed all the dependencies. did you fix spinlock.h? Go into your kernel source directory(or directories if you have more than one kernel source on your system) and edit the file spinlock.h Then goto line 407 Change this line from : #define DEFINE_RWLOCK(x) rw_lock_t x = RW__LOCK_UNLOCKED To: #define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED The problem is that this will have to be done with each new kernel release. If it is an error in the kernel, why does it not just get fixed in any new release of the kernel? What I've done instead is to modify the zaptel Makefile, inserting this single line at line 40: CFLAGS+=$(shell if uname -r | grep -q '2\.6\.9-34.*\.EL'; then echo -Drw_lock_t=rwlock_t; fi) This will fix the problem for any 2.6.9-34 CentOS (or RHEL) kernel. This is a good idea, and worthy of inclusion in the Asterisk tree, IMHO, to stem the steady stream of queries about this problem. However, unless there is a legitimate occurrence of rw_lock_t in other kernel versions, this doesn't really need to be made version-dependent, does it? Could just do this and it would be compatible with any version: CFLAGS+= -Drw_lock_t=rwlock_t Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Setting a threshold for asterisk to take ZAP line off hook ?
On Wed, Jul 19, 2006 at 12:55:56PM +0300, Maxim Vexler wrote: Well, I did managed to get it to work some how with the attached patch. My problems with this code are : Interesting... a. It will destroy all active calls because I'm using zap_restart(). The patch from http://bugs.digium.com/view.php?id=7256 add the function zap_destroy_channel_bynum from the implementation of 'zap destroy channel'. I really need to find a better way to destroy only the active channel instead of the whole zaptel stock. BTW: I also figure that two items from asterisk/include/reply_to_patch_writers.h are: * Please file new patches at the mantis. * New features have better chance at being observed when against trunk * http://www.digium.com/bugguidelines.html I'll look into the problem below later on... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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: CentOS 4.3 and Zaptel-1.2.7
On Wed, Jul 19, 2006 at 10:31:05AM +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Russ Price [EMAIL PROTECTED] wrote: The problem is that this will have to be done with each new kernel release. If it is an error in the kernel, why does it not just get fixed in any new release of the kernel? What is the bug number in the RedHat bugzilla? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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 forwarding to mobile phone
You will need an asterisk server + X100P + GSM Gateway say from cyber-telecom.net You can config the X100P with GSM Gateway like what you would do with an normal Phone line and use it to dial in or out between VoIP and GSM Network From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Mercado Sent: Tuesday, July 18, 2006 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call forwarding to mobile phone I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank sorry for my english, i speek spanish only. atte, Rodrigo M On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote: is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Lito Lampitoc Sent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding to mobile phone Hello all, Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanks Lito ___ --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 ___ --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 forwarding to mobile phone
Yes Get an X100P Sam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc Sent: Tuesday, July 18, 2006 5:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call forwarding to mobile phone is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Lito Lampitoc Sent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding to mobile phone Hello all, Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanks Lito ___ --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: SV: [Asterisk-Users] Nokia E61
WE have found this type of phone work better than E61 http://cyber-telecom.net/shop/product_info.php/cPath/21/products_id/31 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fredrik Emil Jensen Sent: Tuesday, July 18, 2006 4:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: SV: [Asterisk-Users] Nokia E61 Yes. It works at the same time. The problem is the NAT, you will be able to dial in to the phone as long as the NAT table knows where to redirect the packet too, but when the firewall/router loses its table (usually it will timeout after xx sec/min) you will only be able to dial outgoing call. But I see that the phone support TCP, has anyone tried it with SIP TCP through NAT? Or what about running it through the VPN software that is also on the phone? Or what about the http://sofia-sip.sourceforge.net/ has anyone tried that to see if it works with NAT. The phone it's the best SIP / WIFI phone that I have tried, easy to choose which connection (GMS/InternetPhone) you want to dial through and very good sound. For your guys that are planning on using this through hot-spot etc, you can use example use Birdstep Roaming client (http://www.smartroaming.com), and I can also see that iPass is also creating/created a client http://www.ipass.com/pressroom/pressroom_releases.html?rid=201 based on the GRIC stander. This client will log you on the hot-spot automatic. So its only one problem now it's the NAT issue, I guess you can tunnel this traffic, but that's its another client, and more latency! /Fredrik Jensen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devraj Mukherjee Sent: 6. juli 2006 04:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: SV: [Asterisk-Users] Nokia E61 Does the GSM and Wi-Fi phone feature work at the same time? :) Thanks for your time On 7/5/06, Amund Nygaard [EMAIL PROTECTED] wrote: Hello I done some more testing, i have no problems connection behind natted networks. It even connected with 3G, but as you can imagine G711 is not very suited for that :P BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Antonio Rabena Sendt: 5. juli 2006 10:26 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Nokia E61 Hi, I had no issues connecting/calling to my asterisk box (on public ip), even my phone is behind a hotspot. Its just that i need to use G711 codec. At 03:34 PM 7/5/2006, you wrote: Hello Has anyone tried a Nokia E6x phone when it is natted? Like behind a hotspot or similar? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Devraj Mukherjee Sendt: 4. juli 2006 12:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Nokia E61 Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? ___ --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 ___ --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 ___ --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] Issues with MeetMe
Chris Jones wrote: -- Executing MeetMe(SCCP/flat-1-0008, ) in new stack It's a known issue with the CHAN_SCCP driver. There is no fix for it. Install the SIP firmware. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
Bristuff 0.2.0-RC8 if for Asterisk 1.0.10 Bristuff 0.3.0-PRE-1r if for Asterisk 1.2.9.1, libpri 1.2.3 and zaptel 1.2.6 download proper versions (for Asterisk 1.2.9.1) look at install.sh in bristuff directory do as it's written there: cd zaptel patch -p1 ../patches/zaptel.patch cd .. cd libpri-1.2.3 patch -p1 ../patches/libpri.patch cd .. cd asterisk-1.2.9.1 patch -p1 ../patches/asterisk.patch cd .. then try to install Trixbox. [EMAIL PROTECTED] as i kno comes with whole OS and asterisk installation is automated and ther is no time for applying bristuff patches. It looks that You have to manually install OS and asterisk then trixbox -Hope that help You a little. I've been trying to install bristuff on my system for a really long time. This is what I've done so far: I started with a [EMAIL PROTECTED] installation. I tried downloading and compiling bristuff release - it didn't work. It was a long time ago, I don't remamber what the problem was. I tried compiling the latest bristuff (whatever latest was about 1-2 months ago). It failed to compile. I download the full CentOS 4.3 and tried compiling both bristuff release (0.2.0-RC8) and bristuff latest. Again, whatever latest was about 2 weeks ago. Next I found something about bristuff being known to work on kernel 2.4; Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried compiling both bristuff release (0.2.0-RC8) and the current release of today (19 july 2006). I wasn't able to compile ither one of them. Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) and two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling everything a few times, I think it's time to ask for help: Would someone be so kind and tell my how they installed Bristuff from A to Z? (that is, what version of Linux so I can download the same version, what updates, what version of bristuff). I'm hoping for a quick answer like: Install LinuxVariant 10.20, install all updates using LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call install.sh and be done with it. Thanks, Cosmin Prund ___ --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 forwarding to mobile phone
On Wed, 2006-07-19 at 19:04 +0800, Sam Tam wrote: You will need an asterisk server + X100P + GSM Gateway say from cyber-telecom.net Not forgetting that the above person IS cyber-telecom.net. Therefore his advice is not impartial. -- Dave Cotton [EMAIL PROTECTED] ___ --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] -- Going to extension s|1 because of immediate=yes, but immediate is 'no'
We have an asterisk with a TE410P in it, when a call comes in it says :
-- Going to extension s|1 because of immediate=yes
-- Extension 's' in context 'default' from '[calling num]' does not
exist. Rejecting call on channel 0/27, span 2
but in zapata.conf immediate=no :
[channels]
language=it
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
immediate=no
callerid=asreceived
usecallerid=yes
hidecallerid=yes
usecallingpres=yes
so I'm stuck, beacuse if in extension i put s,1,Dial(foobar/${EXTEN}) I
really dial 's' and if I put _X.,1,Dial(foobar/${EXTEN}) I don't even
get there because immediate=yes looks for 's'.
The strange thing is that this configuration works perfectly in other
places, can it be that the connected nortel forces in some way
immediate=yes ?
___
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Re: [asterisk-users] Finding far end echo in Verizon network
Same problem here, the far end has to complain to Verizon for anything to happen unless you want to get an echo canceler. I have know calls routed via a ISTP provider that is known not to echo on Verizon networks. On 7/19/06, carl Lougher [EMAIL PROTECTED] wrote: This is a weird one. Network: Asterisk ver 1-0-9 on DL360. 10 Cisco 7960g phones with 3.8.2 SIP Load. Gateway - Cisco 2811 router with 4 x verizon bri's. Network - Private vlan with 1ms response times to all devices. Issue: Intermittent echo on outbound/inbound calls. Users hearing their own voice about 0.5sec later. Tried so far: Upgraded firmware on some phones to 3.8.2 Upgraded software on Cisco router. Changed gain and attentuation settings on cisco router Got Verizon to test bris Moved rtp from asterisk direct to phone and router (canreinvite=yes) load tested asterisk None of the above made any difference. They are hearing their own voice so that means the issue is on the far end. But should it be up to me to control the possible delay or slippage in the verizon bri network? Any help much appreciated. Taf. ___ All new Yahoo! Mail The new Interface is stunning in its simplicity and ease of use. - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] QueueMetrics 1.2.1 released today
Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones: - Bug fix: the Show members only button was not working in 1.2.0 - Improved graphical layout, gadget sizes and rendering in IE/Firefox - Time zone offsets are now full -24/+24 hours and can be set to a default - It is now possible to reject anomalous calls, i.e. calls where the Asterisk logging went wrong for some reason and are left dangling, forever open. A couple of new properties control the maximum allowed wait and talk time for calls to be considered open. Note: this does not affect calls that are logged correctly, no matter what their length may be. - It is now possible to ignore QUEUESTART events altogether via a configuration option. - New feature: auto-scrolling real-time wallboard. A full list of improvements over version 1.2.0 can be found at http://queuemetrics.loway.it/news.jsp QueueMetrics 1.2.1 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user manual that covers all aspects of it. QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs. You can request a trial key if you run a larger installation and would like to test it in your own environment. The latest version of QueueMetrics can be downloaded from http://queuemetrics.loway.it/download.jsp Hope you like it, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] emulating key system - pick up so and so on line1
Thanks allsounds like a good solution! Lets seecut their phone bill in half and get used to call parkingor continue to pay lots of money. No brainer really! Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, July 18, 2006 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] emulating key system - pick up so and so on line1 Bruce, Good call on this one! Ive found that users can handle small changes if they are parallel with something theyre already comfortable doing. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Tuesday, July 18, 2006 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] emulating key system - pick up so and so on line1 Bill, Our solution was to simply retrain the users to use call parking. The company had used a key system for more then a decade and I thought the change would be a tough one, but for the most part people have handled the change from Pickup line 1 to Pickup 71. Not an exact fit I know, but thought I would offer it since I was in your shoes and have found the transition easier then expected. On 7/18/06, Bill Gibbs [EMAIL PROTECTED] wrote: Is there anyway to use Polycom phones (601, 501s) to emulate a key system where you can have a shared lines that people can pick up instead of using transfer? This would make it easier for users used to putting a call on hold then telling another user so and so is on line 2. I know shared line appearances could do it but obviously that's not supported. Any other suggestions? Bill ___ --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 -- Bruce Nortex Networks ___ --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] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
Thanks for your input. I'm good at following instructions (if I can find instructions) so I'll give anything a try! I'm downloading bristuff from http://junghanns.net/downloads/ and the tar's I'm getting from there contain some kind of bootstraping for the installation. The install.sh file simply calls download.sh and then compile.sh. The download.sh script downloads a specific version of asterisk (and everything else required) so I doubt it gets it wrong. It then patches the thing all by itself, using exactly the instructions you gave. It fails when it tries compiling stuff. I installed CentOS 3 using everything as an install option so I think I've got everything. If you ever got this working, would you be so kind to tell what version of Linux you used and what version of bristuff? I prefer CentOS/Fedora/RHL instalations as that's what I've always used and that's what I know, but I'm willing to use anything as long as it gets the work done. Thanks again, Cosmin Prund Filip Drągowski wrote: Bristuff 0.2.0-RC8 if for Asterisk 1.0.10 Bristuff 0.3.0-PRE-1r if for Asterisk 1.2.9.1, libpri 1.2.3 and zaptel 1.2.6 download proper versions (for Asterisk 1.2.9.1) look at install.sh in bristuff directory do as it's written there: cd zaptel patch -p1 ../patches/zaptel.patch cd .. cd libpri-1.2.3 patch -p1 ../patches/libpri.patch cd .. cd asterisk-1.2.9.1 patch -p1 ../patches/asterisk.patch cd .. then try to install Trixbox. [EMAIL PROTECTED] as i kno comes with whole OS and asterisk installation is automated and ther is no time for applying bristuff patches. It looks that You have to manually install OS and asterisk then trixbox -Hope that help You a little. I've been trying to install bristuff on my system for a really long time. This is what I've done so far: I started with a [EMAIL PROTECTED] installation. I tried downloading and compiling bristuff release - it didn't work. It was a long time ago, I don't remamber what the problem was. I tried compiling the latest bristuff (whatever latest was about 1-2 months ago). It failed to compile. I download the full CentOS 4.3 and tried compiling both bristuff release (0.2.0-RC8) and bristuff latest. Again, whatever latest was about 2 weeks ago. Next I found something about bristuff being known to work on kernel 2.4; Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried compiling both bristuff release (0.2.0-RC8) and the current release of today (19 july 2006). I wasn't able to compile ither one of them. Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) and two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling everything a few times, I think it's time to ask for help: Would someone be so kind and tell my how they installed Bristuff from A to Z? (that is, what version of Linux so I can download the same version, what updates, what version of bristuff). I'm hoping for a quick answer like: Install LinuxVariant 10.20, install all updates using LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call install.sh and be done with it. Thanks, Cosmin Prund ___ --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] Astribank?
On Tue, 18 Jul 2006 22:22:07 -0400, C F wrote: Well tzafrir will know :) Tzafrir here are a few questions: 1. Does the FXO module support: A. Hangup detection? B. Flash? (I'm assuming that yes, since Asterisk sees it as an FXO) C. MWI from telco. D. Dring from telco (again assuming yes, since Asterisk would see that). 2. Can the astribank 16 be condigured with just 8 FXO? or with 16 FXO? 3. What are the configurations available for the asteribank 32? 4. Pricing? Much of what you ask is answered on the Xorcom web site. I am looking for someones specific experience with the products for real-world commentary and observations. Michael ___ --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] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
First question: Do You have kernel sources ? this is required for #make-ing zaptel i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, libpri-1.2.2 and zaptel-1.2.3 OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself so there was kernel sources in system. I didn't use bristuff autamated install. wget-ed asterisk, libpri, zaptel and patched them. there is recomended to use make linux26 when making zaptel on 2.6. kernel. bristuff compile.sh don't have linux26 option Thanks for your input. I'm good at following instructions (if I can find instructions) so I'll give anything a try! I'm downloading bristuff from http://junghanns.net/downloads/ and the tar's I'm getting from there contain some kind of bootstraping for the installation. The install.sh file simply calls download.sh and then compile.sh. The download.sh script downloads a specific version of asterisk (and everything else required) so I doubt it gets it wrong. It then patches the thing all by itself, using exactly the instructions you gave. It fails when it tries compiling stuff. I installed CentOS 3 using everything as an install option so I think I've got everything. If you ever got this working, would you be so kind to tell what version of Linux you used and what version of bristuff? I prefer CentOS/Fedora/RHL instalations as that's what I've always used and that's what I know, but I'm willing to use anything as long as it gets the work done. Thanks again, Cosmin Prund Bristuff 0.2.0-RC8 if for Asterisk 1.0.10 Bristuff 0.3.0-PRE-1r if for Asterisk 1.2.9.1, libpri 1.2.3 and zaptel 1.2.6 download proper versions (for Asterisk 1.2.9.1) look at install.sh in bristuff directory do as it's written there: cd zaptel patch -p1 ../patches/zaptel.patch cd .. cd libpri-1.2.3 patch -p1 ../patches/libpri.patch cd .. cd asterisk-1.2.9.1 patch -p1 ../patches/asterisk.patch cd .. then try to install Trixbox. [EMAIL PROTECTED] as i kno comes with whole OS and asterisk installation is automated and ther is no time for applying bristuff patches. It looks that You have to manually install OS and asterisk then trixbox -Hope that help You a little. I've been trying to install bristuff on my system for a really long time. This is what I've done so far: I started with a [EMAIL PROTECTED] installation. I tried downloading and compiling bristuff release - it didn't work. It was a long time ago, I don't remamber what the problem was. I tried compiling the latest bristuff (whatever latest was about 1-2 months ago). It failed to compile. I download the full CentOS 4.3 and tried compiling both bristuff release (0.2.0-RC8) and bristuff latest. Again, whatever latest was about 2 weeks ago. Next I found something about bristuff being known to work on kernel 2.4; Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried compiling both bristuff release (0.2.0-RC8) and the current release of today (19 july 2006). I wasn't able to compile ither one of them. Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) and two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling everything a few times, I think it's time to ask for help: Would someone be so kind and tell my how they installed Bristuff from A to Z? (that is, what version of Linux so I can download the same version, what updates, what version of bristuff). I'm hoping for a quick answer like: Install LinuxVariant 10.20, install all updates using LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call install.sh and be done with it. Thanks, Cosmin Prund ___ --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: SV: [Asterisk-Users] Nokia E61
On Wed, Jul 19, 2006 at 07:05:55PM +0800, Sam Tam wrote: WE have found this type of phone work better than E61 http://cyber-telecom.net/shop/product_info.php/cPath/21/products_id/31 This is not the .biz list, Sam works for Cyber Telecom !!! So it probably does work better, hm Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Don't Hit # after 9 to get PSTN line
Hi all, Iv got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion The problem occurs when the user doesnt complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo ___ --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] Re: Using dproxy to solve no DNS hangs everything problem?
Yeah, I thought dnsmasq was the cure, too. We had an internet outage last week. It was odd, our ISP (ATT) changed out static IP's (don't ask... No one knows why... At least I figured out what was going on...). Thus, our modem/router was whacked, as well as our firewall. So, I think every piece of hardware saw the link as being up, but of course routing was not operating. For clarity, dnsmasq is installed on our firewall, a PC that routes traffic to our modem/router. As far as I understand, the way dnsmasq works is that it proxys DNS requests to your ISP's DNS servers (or wherever). At some point, if dnsmasq determines that the upstream DNS servers are down/unreachable, then it will respond to DNS requests with a failure/immediately timeout, thus precluding Asterisk from hanging while it waits for a DNS query. Sounds like a good fix for our problem, right? Well, last week it did not behave well. Calls on Zap channels came in, they would sit for several seconds (maybe 20-30, oddly some callers sat through that silence just waiting!), then when no SIP phones could be found it would go to the IVR (all attendants are busy, leave a message, etc...). All connectivity with the SIP phones was gone, even trying to initiate a call from the phones (Cisco _79[46]0's) to the Asterisk server was out. All connectivity on the local LAN was good, even dnsmasq serving the local DNS names. Can anyone explain why Asterisk would not ring the SIP phones on our LAN? The phones all register with the Asterisk server, the Asterisk server does not need to look anything up to contact the IP's of the SIP phones on the local LAN. I just don't know which direction to turn here. Thanks Brian for your work, I have had the same problem I installed dnsmasq and I *think* the problem is gone now, I'm repeating I think, I'll only know when the internet goes down again. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.com ___ --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] QueueMetrics 1.2.1 released today
How to upgrade? Sounds like some great new features and I am just now getting ours fully setup. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lenz Sent: Wednesday, July 19, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] QueueMetrics 1.2.1 released today Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones: - Bug fix: the Show members only button was not working in 1.2.0 - Improved graphical layout, gadget sizes and rendering in IE/Firefox - Time zone offsets are now full -24/+24 hours and can be set to a default - It is now possible to reject anomalous calls, i.e. calls where the Asterisk logging went wrong for some reason and are left dangling, forever open. A couple of new properties control the maximum allowed wait and talk time for calls to be considered open. Note: this does not affect calls that are logged correctly, no matter what their length may be. - It is now possible to ignore QUEUESTART events altogether via a configuration option. - New feature: auto-scrolling real-time wallboard. A full list of improvements over version 1.2.0 can be found at http://queuemetrics.loway.it/news.jsp QueueMetrics 1.2.1 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user manual that covers all aspects of it. QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs. You can request a trial key if you run a larger installation and would like to test it in your own environment. The latest version of QueueMetrics can be downloaded from http://queuemetrics.loway.it/download.jsp Hope you like it, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
Filip Drągowski schrieb: First question: Do You have kernel sources ? this is required for #make-ing zaptel i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, libpri-1.2.2 and zaptel-1.2.3 OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself so there was kernel sources in system. I didn't use bristuff autamated install. wget-ed asterisk, libpri, zaptel and patched them. there is recomended to use make linux26 when making zaptel on 2.6. kernel. bristuff compile.sh don't have linux26 option that linux26 stuff is as far as i know only important to ztdumm.ko, a kernel module which is needed, if you have no Zaptel Cards in your PC and want to use MeetMe Conferencing system. you dont need to tell zaptel wheter you have a 2.6 or 2.4 Kernel, the Makfile discovers this himself. so, no need to worry about 2.4 or 2.6 stuff. Getting kernel sources was a torture for me on Cent-OS 4. maybe somebody can explain how to get them the right way!!! and apply the patches and that. Which Cards do you wanna use in your asterisk (especiallly which ISDN cards, if any) can you post the errormessage of the bristall install script? regards Kai ___ --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] Don't Hit # after 9 to get PSTN line
Turn off 3-way calling on your SIP device. Set the dialplan on your SIP device to not wait 15 seconds after pressing 9. Pablo Mora wrote: Hi all, Iv' got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion The problem occurs when the user doesn't complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Re: Don't Hit # after 9 to get PSTN line
You must have other dialplan entries that start with 9. How does asterisk know you are dialing "9" or one of your other dialplan entries that starts with "9"? I has to wait for the digit timeout. I am curious what this "9" is used to connect to? Are you trying to get dialtone from another PBX? -- -- Steven http://www.glimasoutheast.org "Pablo Mora" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... Hi all, Iv got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion The problem occurs when the user doesnt complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Ztdummy
What I find interesting is that timing will work. However I don't feel
comfortable letting the client use the system if this can affect him in anyway.
Thanks.
Do you have any Zaptel card in the box?
GotoIf($[${ANSWER}=YES]?s-yes|1:s-no|1)
s-yes:
you dont need ztdummy
s-no:
does /dev/zap exist?
maybe some issues with devfs/udev and dabian?
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Re: [asterisk-users] BudgeTone BT-102 not registering to Asterisk
Andrea Spadaccini wrote: Hello everyone, is there any known bug in the inner working of BT-10X series? It doesn't register to one of the testing Asterisk servers that I have at work, I don't understand why!! The logs simply say that it's sending SIP message 1 to the Asterisk Server: Jul 19 11:44:50 192.168.1.40 GS_LOG: [00:0B:82:09:5D:81][000][FFFB][01000817] Send SIP message: 1 To 192.168.1.2:5060 repeated N times (with N 10). The phone is configured correctly, with the username and password specified in sip.conf. Did someone encounter this issue? Nope, works fine with 1.2.10, etc. Best look a little closer at your config's. ___ --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] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
On Wed, Jul 19, 2006 at 03:17:15PM +0200, Filip Drągowski wrote: First question: Do You have kernel sources ? this is required for #make-ing zaptel i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, libpri-1.2.2 and zaptel-1.2.3 OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself so there was kernel sources in system. I didn't use bristuff autamated install. wget-ed asterisk, libpri, zaptel and patched them. there is recomended to use make linux26 when making zaptel on 2.6. kernel. bristuff compile.sh don't have linux26 option If you use Debian, you'd probably be better off with the bristuff asterisk debs. They get automatically built for Sarge as well... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] SOLVED: QuadBRI + TDM + GSM hangup problems
On jul/19/2006, Paco Brufal wrote: The system can receive and make calls perfecty, via ISDN and GSM. But when I configure Asterisk to redirect the calls from ISDN to a mobile telephone (via the GSM modules), when the mobile phone answers, the call is terminated: The solution: Answer() before Dial() to mobile phone O:) -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --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] Re: Don't Hit # after 9 to get PSTN line
Really dont. Dialplan is very simple, please take a look [incoming] exten = s,1,Answer exten = s,2,Background(prueba-pbx) exten = s,3,Set(TIMEOUT(response)=5) exten = 1001,1,Dial,SIP/1001|20 exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial,SIP/1002|20 exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial,SIP/1003|20 exten = 1003,2,Hangup [sip] include = out exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial(SIP/1002,20) exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial(SIP/1003,20) exten = 1003,2,Hangup [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion And yes, Im trying asterisk behind and Ericsson MD110 PBX, and when I hit 9 I ask for an internal line and re-send 9 to get an external line. Thanks Pablo ___ --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] Don't Hit # after 9 to get PSTN line
I really dont understand what you say. Ive been searching in my SIP device (Innomedia 3308), and there isnt any option to disable 3-way calling. Do you refer to sip.conf??? Pablo ___ --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] Ztdummy
Did this error started showing after you updated the kernel through
yum -y update?
On 7/19/06, Kai Ober [EMAIL PROTECTED] wrote:
What I find interesting is that timing will work. However I don't feel comfortable letting the client use the system if this can affect him in anyway. Thanks.
Do you have any Zaptel card in the box?GotoIf($[${ANSWER}=YES]?s-yes|1:s-no|1)s-yes: you dont need ztdummys-no: does /dev/zap exist? maybe some issues with devfs/udev and dabian?
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Re: [asterisk-users] Astribank?
Interesting I wasn't able to find answer to any of the questions in 1. Only some confusing answer to 2, no answer whatsoever for 3, the website just says that it could be mixed. No pricing on the site either. On 7/19/06, Michael Graves [EMAIL PROTECTED] wrote: On Tue, 18 Jul 2006 22:22:07 -0400, C F wrote: Well tzafrir will know :) Tzafrir here are a few questions: 1. Does the FXO module support: A. Hangup detection? B. Flash? (I'm assuming that yes, since Asterisk sees it as an FXO) C. MWI from telco. D. Dring from telco (again assuming yes, since Asterisk would see that). 2. Can the astribank 16 be condigured with just 8 FXO? or with 16 FXO? 3. What are the configurations available for the asteribank 32? 4. Pricing? Much of what you ask is answered on the Xorcom web site. I am looking for someones specific experience with the products for real-world commentary and observations. Michael ___ --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
[asterisk-users] Re: Don't Hit # after 9 to get PSTN line
I agree with Eric, that it must be the local dialplan on the SIP device. -- -- Steven http://www.glimasoutheast.org "Pablo Mora" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... Really dont. Dialplan is very simple, please take a look [incoming] exten = s,1,Answer exten = s,2,Background(prueba-pbx) exten = s,3,Set(TIMEOUT(response)=5) exten = 1001,1,Dial,SIP/1001|20 exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial,SIP/1002|20 exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial,SIP/1003|20 exten = 1003,2,Hangup [sip] include = out exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial(SIP/1002,20) exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial(SIP/1003,20) exten = 1003,2,Hangup [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion And yes, Im trying asterisk behind and Ericsson MD110 PBX, and when I hit 9 I ask for an internal line and re-send 9 to get an external line. Thanks Pablo ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] QueueMetrics 1.2.1 released today
There is an updating.txt file in the web app under WEB-INF/README In practice, updating from 1.2.0 means changing the webapp and keeping the same database. It's really a 5 minutes operation. If you istead used yum to install, just type yum update queuemetrics and you should have it updated in minutes (see the FAQ page at http://queuemetrics.loway.it/faq.jsp for other information on how to install using yum). Hope this helps l. On Wed, 19 Jul 2006 15:46:23 +0200, Steven Totaro [EMAIL PROTECTED] wrote: How to upgrade? Sounds like some great new features and I am just now getting ours fully setup. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lenz Sent: Wednesday, July 19, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] QueueMetrics 1.2.1 released today Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones: - Bug fix: the Show members only button was not working in 1.2.0 - Improved graphical layout, gadget sizes and rendering in IE/Firefox - Time zone offsets are now full -24/+24 hours and can be set to a default - It is now possible to reject anomalous calls, i.e. calls where the Asterisk logging went wrong for some reason and are left dangling, forever open. A couple of new properties control the maximum allowed wait and talk time for calls to be considered open. Note: this does not affect calls that are logged correctly, no matter what their length may be. - It is now possible to ignore QUEUESTART events altogether via a configuration option. - New feature: auto-scrolling real-time wallboard. A full list of improvements over version 1.2.0 can be found at http://queuemetrics.loway.it/news.jsp QueueMetrics 1.2.1 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user manual that covers all aspects of it. QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs. You can request a trial key if you run a larger installation and would like to test it in your own environment. The latest version of QueueMetrics can be downloaded from http://queuemetrics.loway.it/download.jsp Hope you like it, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Don't Hit # after 9 to get PSTN line
No. In SIP these features are configured on the SIP device. If you cannot disable three-way calling, or modify the dialplan on your SIP device, then there is nothing you can do to fix the problem. Pablo Mora wrote: I really don't understand what you say. I've been searching in my SIP device (Innomedia 3308), and there isn't any option to disable 3-way calling. Do you refer to sip.conf??? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] intel vs amd motherboards
i don't think there is ANY difference with 1 or 2 SATA HDD. however here is my single proc Xeon2.8 (512k) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 6 9 -14 ulaw - 3 - 1 2 2 1 6 9 -14 alaw - 3 1 - 2 2 1 6 9 -14 g726 - 3 2 2 - 2 1 6 9 -14 adpcm - 3 2 2 2 - 1 6 9 -14 slin - 2 1 1 1 1 - 5 8 -13 lpc10 - 4 3 3 3 3 2 -10 -15 g729 - 4 3 3 3 3 2 7 - -15 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 710 - - and here is a dual Xeon3.2(1M) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 4 9 -14 ulaw - 2 - 1 2 2 1 4 9 -14 alaw - 2 1 - 2 2 1 4 9 -14 g726 - 2 2 2 - 2 1 4 9 -14 adpcm - 2 2 2 2 - 1 4 9 -14 slin - 1 1 1 1 1 - 3 8 -13 lpc10 - 3 3 3 3 3 2 -10 -15 g729 - 2 2 2 2 2 1 4 - -14 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 510 - - the conclusion to me, is comparing transcoding capabilities with show translation is like bogoMIPS... I have recently build 2 machines, one with an Intel Pentium Dual Core CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2 HDDs. Here are the show translations from both: Intel Dual Core machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 517 -17 ulaw - 2 - 1 2 2 1 517 -17 alaw - 2 1 - 2 2 1 517 -17 g726 - 2 2 2 - 2 1 517 -17 adpcm - 2 2 2 2 - 1 517 -17 slin - 1 1 1 1 1 - 416 -16 lpc10 - 3 3 3 3 3 2 -18 -18 g729 - 4 4 4 4 4 3 7 - -19 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 618 - - AMD 64 bit machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 313 -12 ulaw - 3 - 1 2 2 1 313 -12 alaw - 3 1 - 2 2 1 313 -12 g726 - 3 2 2 - 2 1 313 -12 adpcm - 3 2 2 2 - 1 313 -12 slin - 2 1 1 1 1 - 212 -11 lpc10 - 3 2 2 2 2 1 -13 -12 g729 - 4 3 3 3 3 2 4 - -13 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 414 - - This shows that the AMD 64 bit is worth much more than just the price difference. On 7/6/06, Andrew Kirch [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Don Sent: Wednesday, July 05, 2006 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] intel vs amd motherboards If you want to handle, lets say 1000 calls or more at the same time, you should of course use a better processor. In my opinion, it doesn't matter whether you
Re: [asterisk-users] intel vs amd motherboards
As we are talking about pbx boxes (for large office/enterprises/ maybe ITSP?) i think we are using server grade boxes (like hp ml3xx or bigger) I have some servers with fan on cpu heatsink, but most of them are using only heatsink on cpu, and redundant fans. I think, we need some real life comparison to decide, what to choose. i'm not a cpu expert, but who knows, if dual amd is better for transcoding or dual xeon? i think it can as big weight on paralellisation as big weight on horsepower also, don't you think? Another thing, is what to choose? another cpu (so go for dual, or quad) or bigger cache inside? (probably another 3.2G/1M xeon would cost less, than replace the existing with a 3.2G/2M) So i would welcome (and maybe pay for) a real life test what says: AMD opteron will do x paralell alaw-g.729 dual opteron, fx66 and the same for intel pentium extreme, duo core, xeon with 512k cache, xeon with 1M cache, and probably with 2. and also Xeon DP. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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
[asterisk-users] Callback: Dial(dummy) 10 seconds rining without costs?
Salve *! What is the trick to let the caller hear 10 seconds free-ringing sound and then the busy signalisation of his telco without costs for him? exten = sip1/Unknown,1,Wait(10) exten = sip1/Unknown,2,Hangup Will not create a free-ringing, but: exten = sip1/Unknown,1,Dial(SIP/hardwarephone,10) exten = sip1/Unknown,2,Hangup How do I create a dummy for the ring-signalisation? Ringing(10) would pickup and create costs for the caller. BTW what does Answer do exactly? I like to avoid that the caller has to pay. Answer=Cost for the Caller? And are more then the free, busy signalisation that I could activate with SIP and asterisk? Something like person you called is temporary... or number you called does not exist...? Greetings, rob ___ --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] Problem with MFCR2
Carlos. Unblocking the remote side is NOT your responsibility, unless you own the 2 end points :). I suppose you are getting connected to some telco (avantel, telmex, etc), if so, is telco's responsibility to unlock their side. To discard any problem with Asterisk, try using testcall utility included with libunicall. testcall will allow you to debug the problem without using Asterisk at all. Here is a link on my blog for a document I wrote that may help you out. ( is in spanish, since most people using MFCR are either on Brazil or Mexico ) http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-asterisk-unicall.pdf It describes some debugging techniques. Best Regards On 7/18/06, Carlos Chavez [EMAIL PROTECTED] wrote: I just finished installing Asterisk 1.2.10, Zaptel 1.2.7 on a Centos 4.3 64 bit server. I installed all the mfcr2 requirements and Asterisk seems to be running fine. Now I cannot get unicall to unblock the remote side of the connection. I see the message that the local end has been unblocked but when I dial the DID I get a busy tone and I cannot dial out. I am using spandsp0.0.2pre21, libunicall-0.0.3, libsupertone-0.0.2 and libmfcr2-0.0.3. These are the same exact versions I use on a different server that works perfectly. Basically the only difference is that I am using the newest version of Asterisk and Zaptel and that the server is in another city. Here is the log for asterisk with the unicall log al 255: == Parsing '/etc/asterisk/unicall.conf': Found Loading protocol mfcr2 Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel gains -- Registered channel 1, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Channel gains -- Registered channel 2, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/3 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/3 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/3 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/3 Channel gains -- Registered channel 3, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/4 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/4 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/4 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/4 Channel gains -- Registered channel 4, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/5 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/5 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/5 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/5 Channel gains -- Registered channel 5, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/6 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/6 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/6 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/6 Channel gains -- Registered channel 6, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/7 Call control(8) Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/7 Unblock Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/7 1001 - [1/4000/Idle /Idle ] Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/7 Channel gains -- Registered channel 7, mfcr2 signalling Jul 18 20:27:51 WARNING[21349]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/8 Call control(8) Jul 18 20:27:51
Re: [asterisk-users] Astribank?
Hi I figure I'm technically someone who has first-hand experience with the Astribanks :-) On Tue, Jul 18, 2006 at 10:22:07PM -0400, C F wrote: Well tzafrir will know :) Tzafrir here are a few questions: 1. Does the FXO module support: A. Hangup detection? What type of hangup-detection do you refer to, exactly? B. Flash? (I'm assuming that yes, since Asterisk sees it as an FXO) Yes (Asterisk does the work) C. MWI from telco. What type of MWI? Changing of a dialtone is passed as-is to chan_zap. D. Dring from telco (again assuming yes, since Asterisk would see that). Passed as-i to chan_zap , again. 2. Can the astribank 16 be condigured with just 8 FXO? or with 16 FXO? The available configurations are: 2 FXS (16 FXS ports) 1 FXS + 1 FXO (8 FXS ports, 8 FXO ports) 3. What are the configurations available for the asteribank 32? 4 FXS (32 FXS ports) 3 FXS + 1 FXO (24 FXS ports, 8 FXO ports) 2 FXS + 2 FXO (16 FXS ports, 16 FXO ports) 3 FXS + 1 FXO (24 FXS ports, 8 FXO ports) 4. Pricing? Contact [EMAIL PROTECTED] -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Problems after upgrade asterisk
Hi people, When a I upgrade my asterisk 1.2.4 to asterisk 1.2.9.1 or to asterisk 1.2.10, app_queue, after some time up, doesn't work (I think) When I call to the queue, the channels up: Zap/1-1 [EMAIL PROTECTED]:4 Up Queue(suporte3600) but nothing happens, the asterisk doesn't call any agent (agents are dynamics and they are logged by agentcallbacklogin), i think so because asterisk doesn't spawn a new channel of type LOCAL to call to the agent. My system is in production, so when this problem occurs, my E1 fill up and I type 'show channels' on asterisk console, there are a lot of channels executing app_queue like the line above, but after typed 'show queues' nothing more happened, all commands don't do anything, asterisk shows nothing. All others functions works well, so when this problem occurs I can call normally. I have switched back to asterisk 1.2.4 because this version works well. my system: libpri 1.2.3 zaptel 1.2.7 pentium 4 3.0 GHZ with HT disabled on BIOS TDM2400P with 24 FXS's TE205P with a ISDN E1. all digium cards are in its own irq. How I am doing the upgrade: mv /usr/lib/asterisk /usr/lib/asterisk/old cd asterisk-1.2.10 make install I have tried 'make upgrade' too. -- list of modules loaded is attached. Thanks in advance, and sorry my bad english. -- Iuri Gomes Diniz adm.iuri (at) digi.com.br Network Admin and Programmer [http://clx.digi.com.br] DIGINET [http://www.digi.com.br] Natal - RN - Brazil. Module Description Use Count res_musiconhold.so Music On Hold Resource 1 res_indications.so Indications Configuration0 res_monitor.so Call Monitoring Resource 1 res_adsi.soADSI Resource1 res_agi.so Asterisk Gateway Interface (AGI) 0 res_features.soCall Features Resource 1 res_config_odbc.so ODBC Configuration 1 res_odbc.soODBC Resource0 res_crypto.so Cryptographic Digital Signatures 1 pbx_config.so Text Extension Configuration 0 pbx_spool.so Outgoing Spool Support 1 pbx_loopback.soLoopback Switch 1 pbx_realtime.soRealtime Switch 1 pbx_ael.so Asterisk Extension Language Compiler 0 pbx_functions.so Builtin dialplan functions 0 chan_sip.soSession Initiation Protocol (SIP)1 chan_agent.so Agent Proxy Channel 1 chan_mgcp.so Media Gateway Control Protocol (MGCP)0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_local.so Local Proxy Channel 0 chan_features.so Feature Proxy Channel0 chan_oss.soOSS Console Channel Driver 0 chan_phone.so Linux Telephony API Support 0 chan_zap.soZapata Telephony w/PRI 9 app_dial.soDialing Application 1 app_playback.soSound File Playback Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_directory.so Extension Directory 0 app_mp3.so Silly MP3 Application0 app_system.so Generic System() application 0 app_echo.soSimple Echo Application 0 app_record.so Trivial Record Application 0 app_image.so Image Transmission Application 0 app_url.so Send URL Applications0 app_disa.soDISA (Direct Inward System Access) Appli 0 app_adsiprog.soAsterisk ADSI Programming Application0 app_getcpeid.soGet ADSI CPE ID 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_setcallerid.so Set CallerID Application 0 app_festival.soSimple Festival Interface0 app_queue.so True Call Queueing 2 app_senddtmf.soSend DTMF digits Application 0 app_parkandannounce.so Call Parking and Announce Application0 app_setcidname.so Set CallerID Name0 app_lookupcidname.so Look up CallerID Name from local databas 0
Re: [asterisk-users] Astribank?
Some corrections: 3. What are the configurations available for the asteribank 32? 4 FXS (32 FXS ports) 3 FXS + 1 FXO (24 FXS ports, 8 FXO ports) 2 FXS + 2 FXO (16 FXS ports, 16 FXO ports) 1 FXS + 3 FXO (8 FXS ports, 24 FXO ports) 4. Pricing? Contact [EMAIL PROTECTED] -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Choppy/Jittery playback at beginning of calls
Hi -I am using Asterisk Home 2.7 on a dedicated Linux server to make out going calls. For the most part everything works fine except that most of the voice calls on playback are jittery/choppy at the beginning of the call. After a couple of seconds of choppyness the rest of the message plays back fine. The voice files in question are recorded by the user (as a gsm file)by calling in to the asterisk box over a voicepulse connection. The voice file is then played back via dial plan to designated phone number. I have checked the recorded gsm file and it plays back fine so I know it's not in the recording of the file. I have also checked the format of the gsm file and it is indeed recorded as an 8kb gsm file. When monitoring the outgoing call on IAX2 I can see that it is using the gsm codec, and the iax2.conf file has disallow=all, codec=gsm so gsm should always be forced.I welcome any suggestions as I cannot go to production with this current problem.Thanks. Al. Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --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] Unicall libmfcr
From where did you downloaded the snapshot? could you post a link to the sources? I think this is a problem of missmatch version of old libunicall an newer libmfcr. Those undefined macros should be part of the libunicall headers, so when compiling the new libmfcr2, it does not find the newer libunicall macros. Anyway, im just guessing. I have downloaded what I think is the latest distribution of unicall from soft-switch.org, in download section, and cannot find such macros. Please point at the sources. Regards On 7/19/06, MBIT Technologies [EMAIL PROTECTED] wrote: Hi I'm wondering if you can help me with this error mfcr2.c:3543: error: `UC_REQUESTMOREINFO_ORIGINATING_NUMBER' undeclared (first use in this function) mfcr2.c: In function `call_control': mfcr2.c:3894: error: `UC_OP_REQUESTMOREINFO' undeclared (first use in this function) mfcr2.c:3895: error: `uc_requestmoreinfo_t' undeclared (first use in this function) mfcr2.c:3895: error: syntax error before ')' token make[1]: *** [mfcr2.lo] Error 1 make[1]: Leaving directory `/usr/src/unicall/libmfcr2-0.0.3' make: *** [all] Error 2 I've got the latest snapshot 20060205. Regards Mark Brooker ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Re: Don't Hit # after 9 to get PSTN line
If he modifies the local dialplan on the SIP device, the 3-way issue should go away because he will no longer need to dial a #. Pablo, Look for something like (0T|011x.T|101x.T|x.#|9x.T|*x.T|#xx|393*1x.T|8|5xxx|x11) in the SIP devices config. You would want to change the 9x.T (which means 9 plus any number of digits until timeout, then send the call to asterisk) to just 9 (which means if you dial just a nine, send the call to asterisk) -- -- Steven http://www.glimasoutheast.org Eric ManxPower Wieling [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] No. In SIP these features are configured on the SIP device. If you cannot disable three-way calling, or modify the dialplan on your SIP device, then there is nothing you can do to fix the problem. Pablo Mora wrote: I really don't understand what you say. I've been searching in my SIP device (Innomedia 3308), and there isn't any option to disable 3-way calling. Do you refer to sip.conf??? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
[asterisk-users] Zap channel faxing in or out fails but phone calls work.
I have AAH2.8 on a dual Xeon system with a Sangoma A104 and an Adtran Channel Bank. The system has a single PRI connected to port 1 and port 2 has the T1 cable connected to the Channel Bank. Both are configured properly and work for the inbound/outbound calls and soft-fax reception. I have fax machines connected to FXS ports on the channel bank. The idea here is to allow faxing out over the PRI from these FXS ports and for inbound DIDs to go to specific fax machines. I have [EMAIL PROTECTED] 2.8 setup on this system and I am pretty certain I have it configured correctly. * Each fax machine has it's own Zap extension. * The DID routes to the correct fax machine (zap extension). * I can make and receive phone calls with these fax machines. Meaning, the fax machine has a phone hand set. I can call out with that handset and receive calls on that handset. Here's the problem. When I try send or receive faxes it fails telling me there was a com error. Any ideas? I am at a loss. I have followed the logs. The transmit and receive work. Once the connection is made it fails indicating "com error". Additionally, I hear the "whistle and chirp" of fax machines talking to each other. Combined with being able to make and receive calls over those fax machine hand sets I don't know what the problem is at this point. Please, if anyone has fax machines setup with a similar situation I would appreciate knowing how you have it setup. OS - CentOS 4.3 zaptel - 1.2.5 libpri - 1.2.3 asterisk - 1.2.9.1 freepbx - 2.0.1 Thank you, Greg -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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] Polycom IP301 and Queues
Dean, Thank you for your help. I have it up and running. As soon as I get some free time lets chat about what we need going forward. I have some dollars to move this forward. If I can accommodate additional requirements, all the better. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: Tuesday, July 18, 2006 3:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues The setup looks fine, I will run through what I did and the version, there might be an easier way. cd /usr/src svn checkout http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ asterisk-poly -r 30432 this will checkout the 30432 release and put in the the asterisk-poly directory. cd /usr/src/asterisk-poly make clean make - I found you had to run make (2 or 3 times), it does come up on the screen and tells you to re-run. First run I think makes menuconfig, second can't remember. make mpg123 (if you want mp3 music on hold) make install The only problem I can find in this release is the meetme (conference centre) does not compile, (but ACD does) and in the newer version the meetme works but not ACD. So I'm going to have two servers one for ACD on old software and one for conference on new software. Not great but least it works. Hope that helps. Regards, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Miller Sent: 17 July 2006 23:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues Thanks for the response and information. The Asterisk version that I am using is Asterisk SVN-bweschke-polycom_acd_functions-r37228. I went one revision back using the following command: svn checkout -r37228 http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions PolycomACD-07172006 With no results. I am not as familiar with svn as cvs. I am not sure if the -r option just labels or checks out the requested version. I will do some reading tonight on svn. I have install zaptel and libpri from the latest version of trunk. I am using a Polycom 601 SIP version 1.6.6.0036. The Polycom reg tag includes the following for line button one: reg.1.displayName=Helpdesk reg.1.address=1000 reg.1.label=Agent reg.1.type=private reg.1.thirdPartyName= reg.1.auth.userId= reg.1.auth.password=1000 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires= reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 reg.1.ringType=2 reg.1.lineKeys=1 reg.1.callsPerLineKey=2 I assumed that the property reg.1.auth.userId= is what you meant by not putting in a username on the Polycom. I tried it both ways with no luck. I set the server addrss in the Polycom sip.cfg file. The sip.conf entry for the Polycom looks like: [1000] type= friend secret = 1000 context = default callerid= Helpdesk 1000 accountcode = 1000 host= dynamic nat = no qualify = 1000 canreinvite = no disallow= all allow = ulaw dtmfmode= rfc2833 agentlogin = yes agentcbcontext = default I also have an agent defined in the agnt.conf as: agent = 2000,1234,Test Agent Thanks again for the assistance! Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP301 and Queues I had the same problems, first of all, what version of asterisk are you using? If you run the CLI whats the polycom_acd_functions verison 3. If you did a svn checkout http://polycom_acd_function, then you most likely got the newest version. I had trouble with that. Have you installed and compiled the zaptel/libpri from the trunk? http://svn.digium.com/svn/zaptel/trunk and http://svn.digium.com/svn/libpri/trunk ? You need these for the ACD part. On the polycom setup, make sure the username field is blank and that set a password. In the Sip.conf, make sure the secret is the same as the polycom, and that you do not put a username= or a authname= I can get you all the release/version numbers to download from the svn tomorrow when back in work. It would be easier to talk you through it when in front of the server, but I'm in the UK and the time differences might get in the way! Regards, Dean. - Original Message - From: Michael Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 6:56 PM Subject: RE: [asterisk-users] Polycom IP301 and Queues I have
[asterisk-users] Queue hold position in other language?
Hi, I am wondering how I can change the language of queue hold position. This is probably pretty simple (yes I know I have to record my own soundfiles). What I don't get is where to set the numbers? In queues.conf there are settings for: queue-youarenext = queue-youarenext queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --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] header replacement
On Wed, 2006-07-19 at 16:27 +0800, unplug wrote: I have a header field in the sip message: Remote-Party-ID: [EMAIL PROTECTED] I want to replace it to :Remote-Party-ID: [EMAIL PROTECTED] In Asterisk 1.2 and the trunk, the SipAddHeader and SipGetHeader applications are deprecated. There is a SIP_HEADER dialplan function, instead. You can use it for exactly this purpose. exten = 1234,1,Set(SIP_HEADER(Remote-Party-ID)[EMAIL PROTECTED]) -- Russell Bryant Software Developer Digium, Inc. ___ --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] Zap channel faxing in or out fails but phone calls work.
I had similar problems with a Sangoma card in this configuration. I recently recieved from Sangoma an updated driver that fixed issues with resyncing the clock on the card. You might try getting a hold of Sangoma, David Yat Sin if possible and ask him about it, it may very well be the same problew. On 7/19/06, Gregory L Miller-Kramer [EMAIL PROTECTED] wrote: I have AAH2.8 on a dual Xeon system with a Sangoma A104 and an Adtran Channel Bank. The system has a single PRI connected to port 1 and port 2 has the T1 cable connected to the Channel Bank. Both are configured properly and work for the inbound/outbound calls and soft-fax reception. I have fax machines connected to FXS ports on the channel bank. The idea here is to allow faxing out over the PRI from these FXS ports and for inbound DIDs to go to specific fax machines. I have [EMAIL PROTECTED] 2.8 setup on this system and I am pretty certain I have it configured correctly. * Each fax machine has it's own Zap extension. * The DID routes to the correct fax machine (zap extension). * I can make and receive phone calls with these fax machines. Meaning, the fax machine has a phone hand set. I can call out with that handset and receive calls on that handset. Here's the problem. When I try send or receive faxes it fails telling me there was a com error. Any ideas? I am at a loss. I have followed the logs. The transmit and receive work. Once the connection is made it fails indicating com error. Additionally, I hear the whistle and chirp of fax machines talking to each other. Combined with being able to make and receive calls over those fax machine hand sets I don't know what the problem is at this point. Please, if anyone has fax machines setup with a similar situation I would appreciate knowing how you have it setup. OS - CentOS 4.3 zaptel - 1.2.5 libpri - 1.2.3 asterisk - 1.2.9.1 freepbx - 2.0.1 Thank you, Greg -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --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] Queue hold position in other language?
Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. http://www.voip-info.org/wiki-Asterisk%20sound%20files Hope it helps On 7/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am wondering how I can change the language of queue hold position. This is probably pretty simple (yes I know I have to record my own soundfiles). What I don't get is where to set the numbers? In queues.conf there are settings for: queue-youarenext = queue-youarenext queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Zaptel Compilation Error
On Wed, 2006-07-19 at 18:10 +1000, RR wrote: Think this has been covered several times on the list. Sounds like the spinlock.h issue. You need to go into the kernel directory, for you it seems like the /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/linux/spinlock.h file and replace anywhere you see 'rw_lock' with 'rwlock', make clean, make It just occurred to me that I can easily check for this using autoconf. I have already added autoconf to zaptel in the trunk for other purposes. I'm going to go work on some magic to make it so you never see this problem again from anyone once Asterisk 1.4 is released. :) -- Russell Bryant Software Developer Digium, Inc. ___ --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] Zaptel Problem - Unable to create channel of type 'Zap'
Hello all. I have a Digitum TE110P board configured and working (I think that it's working). When I configure in extensions.conf to a extension route to that board I get "Unable to create channel of type 'Zap'" on log. Here are some configuration: lspci -vv :01:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at c800 [size=256] Region 1: Memory at fc5ff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- ztcfg -vv Zaptel Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. My zaptel.cfg span=1,0,0,cas,hdb3 loadzone=us defaultzone=us bchan=1-15,17-31 dchan=16 My zapata.conf [channels] context=customer signalling=pri_cpe channel = 1 My extensions.conf exten = 4502,1,Dial(Zap/1/4502) show channels in Asterisk CLI: Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) zap show channels in Asterisk CLI: Chan Extension Context Language MusicOnHold pseudo customer 1 customer zap show channel 1 in Asterisk CLI: hannel: 1 File Descriptor: 10 Span: 1 Extension: Dialing: no Context: corsidian Caller ID string: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF PRI Flags: PRI Logical Span: Implicit Actual Hookstate: Onhook When I dial 4502 in the softphone I get "Unable to create channel of type 'Zap'". Could anybody give to me a little help ? Thanks a lot ! Lincoln ___ --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] asterisk core dumps on a Sipura forwarded to a queue/moh
Greetings all,
I'm running Asterisk 1.2.9.1 installed from /usr/ports/net/asterisk on
FreeBSD 6.1-RELEASE.
I'm experiencing a guaranteed asterisk core dump with any Sipura device
set to forward all calls to an extension that is mapped to a queue:
-- Executing Macro(SIP/10040-4c43, call|10027) in new stack
-- Executing Set(SIP/10040-4c43, ext=10027) in new stack
-- Executing Dial(SIP/10040-4c43, SIP/10027|20|o) in new stack
-- Called 10027
-- Got SIP response 302 Moved Temporarily back from 10.20.30.40
-- Now forwarding SIP/10040-4c43 to 'Local/[EMAIL PROTECTED]' (thanks to
SIP/10027-4f37)
sip*CLI
Disconnected from Asterisk server
#
so the 10027 is the Sipura-3000 in this case, with configured Cfwd All
Dest: (forward all calls) to the extension 111, which is a queue or
109, which is a musiconhold call.
-rw--- 1 root wheel 11292672 Jul 19 21:15 asterisk.core
(gdb) bt
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
#2 0x2810a450 in ?? ()
(gdb) bt full
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
No symbol table info available.
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
No symbol table info available.
#2 0x2810a450 in ?? ()
No symbol table info available.
If I set the Cfwd All Dest: in the Sipura configuration interface to a
phone extension (f.e. 10011) everything works ok.
Any clue what's causing this?
--8-- extensions.ael ---8--
context default {
s =Goto(MainIVR|s|1);
};
context Main {
includes {
Gateways;
MainENUM;
};
s =Goto(MainIVR|s|1);
101 = Queue(InfoDesk);
111 = Queue(Support);
121 = Queue(Accounting);
131 = Queue(Admin);
141 = Queue(DomHosting);
};
--8-- extensions.ael ---8--
--8-- queues.conf ---8--
[Support]
timeout=60
context=Main
wrapuptime=15
announce-frequency=30
announce-holdtime=yes
monitor-format=wav49
monitor-join=yes
member = SIP/10061
member = SIP/10062
member = SIP/10063
--8-- queues.conf ---8--
Here is the sip debug:
-- Executing Macro(SIP/10040-681e, call|10027) in new stack
-- Executing Set(SIP/10040-681e, ext=10027) in new stack
-- Executing Dial(SIP/10040-681e, SIP/10027|20|o) in new stack
-- SIP Seeding peer from astdb: '10027' at [EMAIL PROTECTED]:5060
for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as3340d5e5
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 3 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK7776a171;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as0d613efd
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94
Messages-Waiting: no
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 0/1 (0/0)
---
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
We're at 10.20.30.1 port 10656
Video is at 10.20.30.1 port 15268
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311;rport
From: John Doe sip:[EMAIL PROTECTED];tag=as5214182e
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 9210 9210 IN IP4 10.20.30.1
s=session
c=IN IP4 10.20.30.1
t=0 0
m=audio 10656 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 10027
sip*CLI
-- SIP read from 10.20.30.40:5060:
SIP/2.0 200 OK
To: sip:[EMAIL PROTECTED]:5060;tag=31180d12ce1539b5i0
From: Unknown sip:[EMAIL PROTECTED];tag=as3340d5e5
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY,
Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?
On 7/19/06, Maxim Vexler [EMAIL PROTECTED] wrote: On 7/18/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I think when a PSTN line says 'Ring' it's simply for aesthetics... The line is 'answered' the instant * connects to it for two-way audio... (well not that instant but somewhere in the connection process. When you are hearing ringing from the PSTN through a zap card, the rings are coming from the phone company and are just sound. * doesn't decode that and act on it yet.) [snip] Well, I did managed to get it to work some how with the attached patch. My problems with this code are : [snip] Hmmm, I was obviously not aware of the true usage of the Wait() application in the dial plan. Setting Wait(X) before Answer() allowed provides the requested operation with out chan_zap restart. :) Thank you. -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --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
SV: [asterisk-users] Queue hold position in other language?
Okay, thanks! I already have set language to 'se' in indications.conf. Next question. If asterisk where to play a digit - does it look in /sounds/se/digits or /sounds/digits/se ? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta Skickat: den 19 juli 2006 18:12 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Queue hold position in other language? Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. http://www.voip-info.org/wiki-Asterisk%20sound%20files Hope it helps On 7/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am wondering how I can change the language of queue hold position. This is probably pretty simple (yes I know I have to record my own soundfiles). What I don't get is where to set the numbers? In queues.conf there are settings for: queue-youarenext = queue-youarenext queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Zap channel faxing in or out fails but phone calls work.
Just a couple checks... You are using G711u for the FXS - right? Also if possible turn off ECM on the FAX machines Otherwise I have never used Sangoma cars but this configuration works very well with Digium cards, at least with asterisk, I do not use aah On Jul 19, 2006, at 11:11 AM, Bruce Reeves wrote: I had similar problems with a Sangoma card in this configuration. I recently recieved from Sangoma an updated driver that fixed issues with resyncing the clock on the card. You might try getting a hold of Sangoma, David Yat Sin if possible and ask him about it, it may very well be the same problew. On 7/19/06, Gregory L Miller-Kramer [EMAIL PROTECTED] wrote: I have AAH2.8 on a dual Xeon system with a Sangoma A104 and an Adtran Channel Bank. The system has a single PRI connected to port 1 and port 2 has the T1 cable connected to the Channel Bank. Both are configured properly and work for the inbound/outbound calls and soft-fax reception. I have fax machines connected to FXS ports on the channel bank. The idea here is to allow faxing out over the PRI from these FXS ports and for inbound DIDs to go to specific fax machines. I have [EMAIL PROTECTED] 2.8 setup on this system and I am pretty certain I have it configured correctly. * Each fax machine has it's own Zap extension. * The DID routes to the correct fax machine (zap extension). * I can make and receive phone calls with these fax machines. Meaning, the fax machine has a phone hand set. I can call out with that handset and receive calls on that handset. Here's the problem. When I try send or receive faxes it fails telling me there was a com error. Any ideas? I am at a loss. I have followed the logs. The transmit and receive work. Once the connection is made it fails indicating com error. Additionally, I hear the whistle and chirp of fax machines talking to each other. Combined with being able to make and receive calls over those fax machine hand sets I don't know what the problem is at this point. Please, if anyone has fax machines setup with a similar situation I would appreciate knowing how you have it setup. OS - CentOS 4.3 zaptel - 1.2.5 libpri - 1.2.3 asterisk - 1.2.9.1 freepbx - 2.0.1 Thank you, Greg -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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 -- Bruce Nortex Networks ___ --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] Zaptel Problem - Unable to create channel of type 'Zap'
Are you sure the other end is configured properly? What does zttool says? Have you turned on all the asterisk debug messages to look further? Regards On 7/19/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote: Hello all. I have a Digitum TE110P board configured and working (I think that it's working). When I configure in extensions.conf to a extension route to that board I get Unable to create channel of type 'Zap' on log. Here are some configuration: lspci -vv :01:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at c800 [size=256] Region 1: Memory at fc5ff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- ztcfg -vv Zaptel Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. My zaptel.cfg span=1,0,0,cas,hdb3 loadzone=us defaultzone=us bchan=1-15,17-31 dchan=16 My zapata.conf [channels] context=customer signalling=pri_cpe channel = 1 My extensions.conf exten = 4502,1,Dial(Zap/1/4502) show channels in Asterisk CLI: Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) zap show channels in Asterisk CLI: Chan Extension Context Language MusicOnHold pseudocustomer 1customer zap show channel 1 in Asterisk CLI: hannel: 1 File Descriptor: 10 Span: 1 Extension: Dialing: no Context: corsidian Caller ID string: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF PRI Flags: PRI Logical Span: Implicit Actual Hookstate: Onhook When I dial 4502 in the softphone I get Unable to create channel of type 'Zap'. Could anybody give to me a little help ? Thanks a lot ! Lincoln ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [asterisk-users] call forwarding to mobile phone
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam Sent: 19 July 2006 12:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] call forwarding to mobile phone You will need an asterisk server + X100P + GSM Gateway say from cyber-telecom.net You can config the X100P with GSM Gateway like what you would do with an normal Phone line and use it to dial in or out between VoIP and GSM Network From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Mercado Sent: Tuesday, July 18, 2006 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call forwarding to mobile phone I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank sorry for my english, i speek spanish only. atte, Rodrigo M On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote: is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Lito Lampitoc Sent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding to mobile phone Hello all, Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanks Lito Hi Sam, Uhm... Wah? Your saying to call a mobile number you need a gsm gateway? What have you been smoking and where can I get some? Last I heard you can use a standard telephone line.. One of us must be on cloud nine! Steve Daniels -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.1/391 - Release Date: 18/07/2006 ___ --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] Simple But important question (for me)
Hi.I'm 100% newbie (in asterisk)I need to know if i can use astersik for something like this:1- receive the call (obvious)2- get the Caller ID3- Send the CID to another application and get some info from a Database example: Your address is some address 4- Get that info and convert it into voice (by mixing various audio files)5- return it to the Caller (as audio)6- use keypress as menu options menu or confirmation responses (i know asterisk can do this) sorry is that sounds pretty obvious to you, but as I said I'm new on this.after this (if the answer is yes) i will read as much documentation as possible to do the rest by myself.-- -- Papita = papa pequeñaPapota = papa grandePaputa = Papa Gigante ..?-- ___ --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] Zap channel faxing in or out fails but phone calls work.
Jerry Jones wrote: Also if possible turn off ECM on the FAX machines This is unsound advice. Why do you think this could possily help? Lee. ___ --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] Asterisk patches for packetcable
Hi GroupDespite of the limits on Asterisk and PacketCable, I've found this web site where I found patches to * to work with packetcable NCS.http://asterisk.urtho.net/tiki-index.php I know this is a halted project but it give me some hope to make some research and to make work the eMTA from Motorola and *Anyone knows how to plug these patches into * ? someone has experience with this patches? does it works? Thanks for your helpCarlos Bernat ___ --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] Simple But important question (for me)
Camilo Echeverry wrote: 1- receive the call (obvious) 2- get the Caller ID 3- Send the CID to another application and get some info from a Database example: Your address is some address 4- Get that info and convert it into voice (by mixing various audio files) 5- return it to the Caller (as audio) Yes, Asterisk can do all of these. You might want to look at AGI/FastAGI for implementing it. # http://www.voip-info.org/wiki/view/Asterisk+AGI # http://www.voip-info.org/wiki-Asterisk+FastAGI If your primary development language is Java you might also be interested in Asterisk-Java which allows you to easily implement AGI scripts in Java: http://asterisk-java.org =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital signature ___ --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] Simple But important question (for me)
The short answer is yes On 7/19/06, Camilo Echeverry [EMAIL PROTECTED] wrote: Hi. I'm 100% newbie (in asterisk) I need to know if i can use astersik for something like this: 1- receive the call (obvious) 2- get the Caller ID 3- Send the CID to another application and get some info from a Database example: Your address is some address 4- Get that info and convert it into voice (by mixing various audio files) 5- return it to the Caller (as audio) 6- use keypress as menu options menu or confirmation responses (i know asterisk can do this) sorry is that sounds pretty obvious to you, but as I said I'm new on this. after this (if the answer is yes) i will read as much documentation as possible to do the rest by myself. -- -- Papita = papa pequeña Papota = papa grande Paputa = Papa Gigante ..? -- ___ --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] Astribank?
On 7/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Hi I figure I'm technically someone who has first-hand experience with the Astribanks :-) On Tue, Jul 18, 2006 at 10:22:07PM -0400, C F wrote: Well tzafrir will know :) Tzafrir here are a few questions: 1. Does the FXO module support: A. Hangup detection? What type of hangup-detection do you refer to, exactly? Someone correct me if I'm wrong, but I think in the US Polarity Reverse is whats used. B. Flash? (I'm assuming that yes, since Asterisk sees it as an FXO) Yes (Asterisk does the work) C. MWI from telco. What type of MWI? Changing of a dialtone is passed as-is to chan_zap. There is also an ADSI type of MWI used in the US. It comes in as a short ring to turn on the MWI lamp on Analog devices, and comes in as a short ring to turn it off as well. It's the same that Sipura ATA FXS ports use for MWI, and I believe Digium FXS uses as well. D. Dring from telco (again assuming yes, since Asterisk would see that). Passed as-i to chan_zap , again. 2. Can the astribank 16 be condigured with just 8 FXO? or with 16 FXO? The available configurations are: 2 FXS (16 FXS ports) 1 FXS + 1 FXO (8 FXS ports, 8 FXO ports) 3. What are the configurations available for the asteribank 32? 4 FXS (32 FXS ports) 3 FXS + 1 FXO (24 FXS ports, 8 FXO ports) 2 FXS + 2 FXO (16 FXS ports, 16 FXO ports) 3 FXS + 1 FXO (24 FXS ports, 8 FXO ports) So you saying there always has to be an FXS card in there? For both the 16 and 32? Also, is it possible to have the 32 with just 16 ports and add the rest when needed? 4. Pricing? Contact [EMAIL PROTECTED] -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Re: CentOS 4.3 and Zaptel-1.2.7
Tzafrir == Tzafrir Cohen [EMAIL PROTECTED] writes: Tzafrir Tzafrir On Wed, Jul 19, 2006 at 10:31:05AM +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Russ Price [EMAIL PROTECTED] wrote: Tzafrir The problem is that this will have to be done with each new kernel release. If it is an error in the kernel, why does it not just get fixed in any new release of the kernel? Tzafrir Tzafrir What is the bug number in the RedHat bugzilla? https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 -- http://rgr.freeshell.org/ ___ --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] Setting a threshold for asterisk to take ZAP line off hook ?
On Wed, Jul 19, 2006 at 07:21:20PM +0300, Maxim Vexler wrote: On 7/19/06, Maxim Vexler [EMAIL PROTECTED] wrote: On 7/18/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I think when a PSTN line says 'Ring' it's simply for aesthetics... The line is 'answered' the instant * connects to it for two-way audio... (well not that instant but somewhere in the connection process. When you are hearing ringing from the PSTN through a zap card, the rings are coming from the phone company and are just sound. * doesn't decode that and act on it yet.) [snip] Well, I did managed to get it to work some how with the attached patch. My problems with this code are : [snip] Hmmm, I was obviously not aware of the true usage of the Wait() application in the dial plan. Setting Wait(X) before Answer() allowed provides the requested operation with out chan_zap restart. Still, I wonder if it would be possible to make it possible to answer an analog line after a specific number of rings rather than a specified time. Easier o think that way. A WaitRings() application? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Simple But important question (for me)
Use festival text to speech for saying the address - Original Message - From: Camilo Echeverry To: asterisk-users@lists.digium.com Sent: Wednesday, July 19, 2006 12:49 PM Subject: [asterisk-users] Simple But important question (for me) Hi.I'm 100% newbie (in asterisk)I need to know if i can use astersik for something like this:1- receive the call (obvious)2- get the Caller ID3- Send the CID to another application and get some info from a Database example: Your address is "some address" 4- Get that info and convert it into voice (by mixing various audio files)5- return it to the Caller (as audio)6- use keypress as menu options menu or confirmation responses (i know asterisk can do this) sorry is that sounds pretty obvious to you, but as I said I'm new on this.after this (if the answer is yes) i will read as much documentation as possible to do the rest by myself.-- -- Papita = "papa pequeña"Papota = "papa grande"Paputa = Papa Gigante ..?-- ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.1/391 - Release Date: 7/18/2006 ___ --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] Zap channel faxing in or out fails but phone calls work.
On 7/19/06, Lee Howard [EMAIL PROTECTED] wrote: Jerry Jones wrote: Also if possible turn off ECM on the FAX machines This is unsound advice. Why do you think this could possily help? Lee. ___ --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 Actually it's quite rational. Check the paragraph on ECM at http://www.voip-info.org/wiki-Asterisk+fax -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --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] Stuck ACD Agents
I have a problem here, when an ACD agent is stuck in PAUSED mode. As you can see from the outout of 'show queues' below, the agent 80014133 has a status of paused. Why is there a 'not in use' after the paused? hestia*CLI show queues oe_techsupp has 0 calls (max unlimited) in 'rrmemory' strategy (14s holdtime), W:0, C:3, A:10, SL:0.0% within 0s Members: Agent/80014154 (Unavailable) has taken no calls yet Agent/80014109 (Busy) has taken 1 calls (last was 3211 secs ago) Agent/80014150 (Unavailable) has taken no calls yet Agent/80014133 (paused) (Not in use) has taken no calls yet Agent/80014151 (Unavailable) has taken no calls yet Agent/80014152 (Busy) has taken 2 calls (last was 1036 secs ago) Agent/80014157 (Unavailable) has taken no calls yet Agent/80014155 (Unavailable) has taken no calls yet No Callers I just tried to log unpause the agent. The status remained the same. I also logged the agent out and back in again. Still, no change. -- Executing Answer(IAX2/216.187.142.203:4569-8, ) in new stack -- Executing Wait(IAX2/216.187.142.203:4569-8, 1) in new stack -- IAX2/216.187.142.203:4569-7 answered SIP/80014133-a4db -- Executing AgentCallbackLogin(IAX2/216.187.142.203:4569-8, 80014133||) in new stack -- Playing 'agent-pass' (language 'en') -- Playing 'agent-newlocation' (language 'en') -- Playing 'agent-loggedoff' (language 'en') == Callback Agent '80014133' logged out -- Hungup 'IAX2/216.187.142.203:4569-7' -- Executing Answer(IAX2/216.187.142.203:4569-10, ) in new stack -- IAX2/216.187.142.203:4569-9 answered SIP/80014133-c708 -- Executing Wait(IAX2/216.187.142.203:4569-10, 1) in new stack -- Executing AgentCallbackLogin(IAX2/216.187.142.203:4569-10, 80014133||[EMAIL PROTECTED]) in new stack -- Playing 'agent-pass' (language 'en') == Setting global variable 'AGENTBYCALLERID_80014133' to '80014133' -- Playing 'agent-loginok' (language 'en') == Callback Agent '80014133' logged in on [EMAIL PROTECTED] -- Executing Hangup(IAX2/216.187.142.203:4569-10, ) in new stack Could this be a bug? Doug. ___ --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] Zaptel Problem - Unable to create channel of type 'Zap'
Hello Moises. I enabled debug mode in asterisk. When I dial, I get:
Jul 19 14:24:38 DEBUG[4463]: build_route: Contact hop:
sip:192.168.0.6:5060
Jul 19 14:24:38 VERBOSE[4463]: -- Executing
Dial("SIP/192.168.0.6-08137090", "Zap/1/4502") in new stack
Jul 19 14:24:38 NOTICE[4463]: Unable to create channel of type 'Zap'
Jul 19 14:24:38 VERBOSE[4463]: == Everyone is busy/congested at this
time
Jul 19 14:24:38 DEBUG[4463]: Exiting with DIALSTATUS=CHANUNAVAIL.
zttool say:
OK Digium Wildcard TE110P T1/E1 Card 0
Current Alarms: No alarms.
Sync Source: Internally clocked
IRQ Misses: 14
Bipolar Viol: 0
Tx/Rx Levels: 0/ 0
Total/Conf/Act: 31/ 31/ 0
112333
1234567890123456789012345789012
I thing that the other end it's ok because I can close the ISDN Link
between the two machines.
Thanks
Lincoln
Moises Silva wrote:
Are you sure the other end is configured properly?
What does "zttool" says?
Have you turned on all the asterisk debug messages to look further?
Regards
On 7/19/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote:
Hello all. I have a Digitum TE110P board configured and working (I
think
that it's working). When I configure in extensions.conf to a extension
route
to that board I get "Unable to create channel of type 'Zap'" on log.
Here are some configuration:
lspci -vv
:01:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN
interface
Subsystem: Unknown device 795e:0001
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop-
ParErr-
Stepping- SERR+ FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium
TAbort-
TAbort- MAbort- SERR- PERR-
Latency: 64 (250ns min, 32000ns max)
Interrupt: pin A routed to IRQ 11
Region 0: I/O ports at c800 [size=256]
Region 1: Memory at fc5ff000 (32-bit, non-prefetchable)
[size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA
PME(D0+,D1-,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
ztcfg -vv
Zaptel Configuration
==
SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: Individual Clear channel (Default) (Slaves: 24)
Channel 25: Individual Clear channel (Default) (Slaves: 25)
Channel 26: Individual Clear channel (Default) (Slaves: 26)
Channel 27: Individual Clear channel (Default) (Slaves: 27)
Channel 28: Individual Clear channel (Default) (Slaves: 28)
Channel 29: Individual Clear channel (Default) (Slaves: 29)
Channel 30: Individual Clear channel (Default) (Slaves: 30)
Channel 31: Individual Clear channel (Default) (Slaves: 31)
31 channels configured.
My zaptel.cfg
span=1,0,0,cas,hdb3
loadzone=us
defaultzone=us
bchan=1-15,17-31
dchan=16
My zapata.conf
[channels]
context=customer
signalling=pri_cpe
channel = 1
My extensions.conf
exten = 4502,1,Dial(Zap/1/4502)
show channels in Asterisk CLI:
Channel (Context Extension Pri ) State Appl.
Data
0 active channel(s)
zap show channels in Asterisk CLI:
Chan Extension Context Language MusicOnHold
pseudo customer
1 customer
zap show channel 1 in Asterisk CLI:
hannel: 1
File Descriptor: 10
Span: 1
Extension:
Dialing: no
Context:
Re: [asterisk-users] Queue hold position in other language?
didn't test it , but i think it will be /sounds/se/digits , setting Language to se will point to /sounds/se and then asterisk will keep the same logic as per default is sounds directory. This is a guess, please test it. On 7/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Okay, thanks! I already have set language to 'se' in indications.conf. Next question. If asterisk where to play a digit - does it look in /sounds/se/digits or /sounds/digits/se ? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta Skickat: den 19 juli 2006 18:12 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Queue hold position in other language? Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. http://www.voip-info.org/wiki-Asterisk%20sound%20files Hope it helps On 7/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am wondering how I can change the language of queue hold position. This is probably pretty simple (yes I know I have to record my own soundfiles). What I don't get is where to set the numbers? In queues.conf there are settings for: queue-youarenext = queue-youarenext queue-thereare = queue-thereare ..but no settings for one, two, three and so on. How do I do this? Do I have to overwrite the default files (which I don't want to do)? Regards, Jan ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Can't get blind transfer to work
Hi, Now that i fixed the problem with roundrobin, now i can't get Blind Transfer to work. I already tried to modify blindxfer option in features.conf with almost any number and still doesn't work. When i dial an extension. I pick up the phone, and then i press # to transfer the call and nothing happens, i can hear the # tone in the other phone. Somebody had the same problem? I need to do a blind transfer in order to do a conference. Anyone has any other options or conference config? i'm trying to follow this instrucions: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macroview_comment_id=11271 but i can't continue if Blind Transfer doesn't work :( Cheers! Santiago ___ --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] Zap channel faxing in or out fails but phone calls work.
without ecm - line errors will cause slight imperfections (dots) on transmitted image with ecm - retry, retry, retry, fail On Jul 19, 2006, at 12:23 PM, Maxim Vexler wrote: On 7/19/06, Lee Howard [EMAIL PROTECTED] wrote: Jerry Jones wrote: Also if possible turn off ECM on the FAX machines This is unsound advice. Why do you think this could possily help? Lee. ___ --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 Actually it's quite rational. Check the paragraph on ECM at http://www.voip-info.org/wiki-Asterisk +fax -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --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
