[asterisk-users] New release of billing and routing software MOR
Hello, We are glad to announce new release of our advanced billing and routing package for Asterisk - MOR v0.7 It is complete solution for VoIP billing and routing for advanced and start-up telecoms, carriers, voip calling card operators and ISPs. Demo available online, as LiveCD or as InstallCD. Contact us for more details. More info: http://www.kolmisoft.com What is new in this version: * Call Routing by priority (Manual LCR) * LCR/Tariff change based on call prefix * PBX Functions - small functions which extends functionality of MOR PRO * PDF UTF8 support * More statistical data * New permission system * Accountant role * CallerID Manipulation: * Localization/Provider Rules * CallerID change on Forward * SIP debug system * New payment gateways: LinkPoint and CyberPlat * Google Maps integration to show Active Calls on the map!!! * IVR system * Limit calls per provider/did/user/device basis * User/Device/DID import from files * Send invoices by email in batches * NO ANSWER/BUSY interpretation for providers * Currency engine rework - automatic update from web Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM gateways - which one ?
I recommand what I know and I sell. Vierling Ecotel VoIP for small capacities and Vierling VTM-pro for ISDN interfaces Regards, Hakem, 2008/11/29 Michael Graves [EMAIL PROTECTED] Portech makes larger rack mounted modular multi-channel gateways as well. Not sure about the ISDN interface, but certainly with T-1/E-1 PRI. Michael On Sat, 29 Nov 2008 14:57:02 +, Julian Lyndon-Smith wrote: Thanks Gordon, I have been playing with the Portech, but was wanting a larger solution (20+ channels) Julian. Gordon Henderson wrote: On Sat, 29 Nov 2008, Julian Lyndon-Smith wrote: I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a quotation. Have had good results with Porech ones Guessing you're in the UK from whois on the domain name, so: £130: http://www.voipon.co.uk/portech-gsm-gateways-portech-voip-gsm-gateway-c-3_192_193.html or £125: http://www.discountphonesystems.co.uk/acatalog/Portech_VoIP_GSM_Gateways.html Ethernet+SIP in, GSM out... (Wait until Monday when the VAT rate drops ... I bet this weekend is going to be a pi$$ poor shopping weekend!!!) (and I don't work for either those companies, just use them for hardware) Gordon __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Hakem Voip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cepstral vs festival
I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which installation policy is behind Asterisk doc delivered with source code ?
On Tue, Dec 02, 2008 at 11:17:11AM +0100, Olivier wrote: 2008/12/2 Tzafrir Cohen [EMAIL PROTECTED] On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote: Hi, Testing latest 1.6.1, it occurred to me I had to add a couple of noload statements in /etc/asterisk/modules.conf to remove ERROR messages, when starting Asterisk. (I don't imply those ERROR messages were fatal to Asterisk but as a general rule, I tried to start Asterisk without any of those). Could you be specific? I think that some messages may be wrongly-labeled. From memory, I had for instance Failed to open /dev/dahdi/transcode: No such file or directory on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. /dev/dahdi/transcode should only be required for codec_dahdi.so . Most people don't need it and would be annoyed by the warning[0] Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with check hardware compliance. So maybe, the policy is to have a somehow functional system with any of mentioned compliant hardware, with all enhancements such as running as non-root, set apart. As I'm not very familiar with module concepts at the moment, I told myself it would be helpful, if strictly following instructions included in README files, I did get a system that starts without any ERROR message of any kind and still could provide some basic telephony services. So my question is : - is there a policy fixing the target of README files ? Yes, please submit fixes. - for example, if someone installs Asterisk (according README files) on a platform equipped with a Digium analog board, should this board be automatically discovered, configured and ready to run ? Not yet: # which modules to load: a temporary workaround That's the point : if optimizing modules load could be postponed to a later stage, that would be better, IMHO, as installing from source is already a long process. modules here referes to DAHDI (kernel) modules rather than Asterisk ones. It refers to an inherent limitation of Zaptel/DAHDI - the order of hardware discovery sets the order of channels. However fixing this is not trivial to say the least. So let's leave it aside for now. It's not really something you care about if you just have a single card, anyway :-) # dahdi_modules is a simple two-liner scrippt that is currently not # installed by default. I figure I should get that functionality added # to dahdi_genconf dahdi_modules /etc/dahdi/modules /etc/init.d/dahdi start # generate /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf dahdi_genconf dahdi_cfg # edit chan_dahdi.conf accordingly. e.g.: echo '#include dahdi-channels.conf' /etc/asterisk/chan_dahdi.conf # apply changes: start/restart asterisk, or: asterisk -rx 'dahdi restart' IMHO, setting a policy would help to guide efforts of many people involved. That done, maybe we would conclude an interactive script would be the missing piece to incorporate user choices that are hard to default to. The above procedure is not a policy. The fact that a configuration can be generaed automatically without manual fine tuning indicates to me that it should be. Currently I'm looking into ways of doing so (and still allowing simple manual overrides). I hope that generally running dahdi_cfg on each span separately on each span at post-registration would do the trick, but this still takes exposing information through sysfs or whatever. As a side note, I would rather avoid interactive setup scripts as they tend to be complicated to automate and force you to keep feeding the same selections again and again. Sangoma's setup script is, IMHO, an example of an interactive script that requires an expert, and takes a heaps of screen space. Not to mention changes a half the system to fit its grand plan. [0] Either actively annoyed by it, or learn to ignore it, and hence later on ignore a warning / error you should have read. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Persistentmembers (Not working with restart)
Hello All, I currently have an Asterisk Box, running a callcenter with 04 queues. I set queues.conf with persistentmembers=yes in the general section as follows: [general] monitor-type = MixMonitor persistentmembers = yes However when I perform any kind of restart in the Asterisk application, all agents are considered unavailable after that. Though when performing reload, agents keep their status as it was before the reload. Is there any where else that I should set dynamic agents as persistent members to keep their status after a asterisk restart?? Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Packets
Dear Sir, My Asterisk server is sending periodically the below SIP packets Retransmitting #4 (NAT) to 68.62.168.138:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK37f337ed;rport From: asterisk sip:[EMAIL PROTECTED];tag=as078bf319 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 02 Dec 2008 12:30:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS [Dec 2 12:30:24] WARNING[6669]: rtp.c:891 ast_rtcp_read: RTCP Read too short Reliably Transmitting (NAT) to 68.62.168.138:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK34e06ad8;rport From: asterisk sip:[EMAIL PROTECTED];tag=as141b747c To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 02 Dec 2008 12:30:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 I checked my system and found out that there is an extension (Type=friend) where ipaddr=68.62.168.138...I removed the ipaddr from the definition of the extension, reloaded the Asterisk but it still sending SIP packets... Please let me know how to force my asterisk from sending such packets and if someone has an idea about similar problem Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 18:56 12/1/2008, Tilghman Lesher wrote: On Monday 01 December 2008 06:21:33 pm Doug wrote: We tell our customers that they are not allowed to download copyrighted material. So your customers are only allowed to download public domain material? That kind of restricts the amount of information available on the Internet. Nitpick: just about everything, including this email, is copyrighted by somebody. Forbidding the download of copyrighted works is not only a draconian policy, but may actually violate several copyright laws (you're interfering with a copyright owner's right to distribute his/her/their works, and courts are generally not very sympathetic with your position). Oops! Didn't mean to start a fire here. I meant to say illegal copyrighted material. Also, if they are using up hundreds of Internet connections, we can see that. It essentially causes a Denial of Service situation for other users on that leg of our wireless network. The system supposedly has rate limiting, but seems to get overloaded when someone goes completely nuts with BitTorrent. We are working on ways to limit the number of simultaneous connections. When we get a copyright infringment notice from our upstream provider, we are compelled to reprimand the user. I don't think we have sent a customer to the shower even if they had several notices. Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Es tut mir leid für das Durcheinander meine Brüder! This is the classic logical fallacy that people seem to perpetuate when reporting news about P2P activity. ISPs oversubscribe. It's a common practice, and reasonably valid. But when you oversubscribe, you use a model based on 'projected' use of the available circuits and bandwidth. If you have a user who pays for a circuit that you've advertised as an X Mb line, and he uses X Mb ALL the time, he's using what he's paying for. If you then proceed to tell him that he can't do that, you're either wrong or you're not being up front enough with your pricing and marketing materials. You can't then proceed to blame the customer for use you did not anticipate. Imagine a farmer who sells tomatoes. He's promised you a bushel, but he gets a harvest of only so many. You walk up to the counter just after he's sold all of his tomatoes to someone and he tells you Sorry. There are no more tomatoes because that customer before you just 'stole' them all from you. He's abusing his privileges by buying up my whole crop. Now whose fault is it that you don't get the tomatoes you want? Is it the customer's fault for buying all the tomatoes the farmer sold him? Or is it the farmer's fault for selling them? The same works with the ISP vs P2P argument. If the ISPs were up-front about saying that they do not intend for you to actually USE the bandwidth you think you're paying for, I would say they had a leg upon which to stand. However, hiding this information from the customer and then blaming the customer when he does what he believes is well within his rights... it may play well in the media, but it's bad for the whole system and is incredibly divisive. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking calls
Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on .. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? Thanks __ Information from ESET Smart Security, version of virus signature database 3655 (20081201) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3655 (20081201) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
On December 1, 2008 07:21:33 pm Doug wrote: Hmmm. When our users are pounding the network with BitTorrent traffic, we just shut them down and wait for them to complain. It's against our Acceptable Use Policy, and causes all sorts of VOIP headaches. As someone who is the technical lead for several ISPs, it is my professional opinion that you haven't a clue how to run such a thing. Torrent does not interfere with VOIP on a well-designed network any more than FTP or web browsing. Honestly, hire a competent admin to set up and run your infrastructure. If torrent's killing VOIP, that means that adding more VOIP will also kill it. Or excessive web browsing. Thank God I'm not one of your customers. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Dial M option from extensions.ael
Hi,
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
I tried both (and variations of those with ^ instead of ,) :
Dial(Local/${EXTEN:1},,M(mymacro(${EXTEN}));
Dial(Local/${EXTEN:1},,M(mymacro,${EXTEN}));
both lead to unexpected results:
macro_exec: No such context 'macro-mymacro' for macro 'mymacro'
As a workaround, I could write my macro in extensions.conf, but ...
Regards
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Re: [asterisk-users] OT: What do you guys think of this?
Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get both channel ids from diaplan ?
Hi,
I think this have been talked over several times but I couldn't find any
answer.
Sorry for asking.
I want from dialplan, to transfer a callee to a context-extension-priority
that would play a given fax file to callee (callee is supposed to be a fax
number).
I can get caller's channel id (with built-in CHANNEL variable).
I found BRIDGEPEER but its value remains unset (see bellow) even inside
connect2fax routine (in which I would like to re-direct both channels :
incoming channel to let caller hear you successfully sent a fax and
outgoing channel to get fax content) :
context mylocal {
_2X. = {
NoOp(Calling ${EXTEN:1} from ${CALLERID(num)} using
${CHANNEL});
NoOp(Peer is ${BRIDGEPEER});
Dial(Local/${EXTEN:1},,U(connect2fax,${EXTEN},${UNIQUEID},${BRIDGEPEER}));
Hangup();
};
};
Any idea ?
Cheers
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Re: [asterisk-users] Using Dial M option from extensions.ael
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
I tried both (and variations of those with ^ instead of ,) :
Dial(Local/${EXTEN:1},,M(mymacro(${EXTEN}));
Dial(Local/${EXTEN:1},,M(mymacro,${EXTEN}));
both lead to unexpected results:
macro_exec: No such context 'macro-mymacro' for macro 'mymacro'
I'd say use the U option instead of G.
AEL macros are converted to Gosub routines, not Macros.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Which installation policy is behind Asterisk doc delivered with source code ?
Err... to follow-up just regarding error messages: On Tue, Dec 02, 2008 at 11:17:11AM +0100, Olivier wrote: 2008/12/2 Tzafrir Cohen [EMAIL PROTECTED] On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote: Hi, Testing latest 1.6.1, it occurred to me I had to add a couple of noload statements in /etc/asterisk/modules.conf to remove ERROR messages, when starting Asterisk. (I don't imply those ERROR messages were fatal to Asterisk but as a general rule, I tried to start Asterisk without any of those). Could you be specific? I think that some messages may be wrongly-labeled. From memory, I had for instance Failed to open /dev/dahdi/transcode: No such file or directory on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. /dev/dahdi/transcode should only be required for codec_dahdi.so . Most people don't need it and would be annoyed by the error[0]. Those who actually need it would wander arould for hours if there's no clear message to indicate it is missing. Suppose there wouldn't be such an error. What indication would you have on the CLI to the fact that the dahdi codec is not working? Or more specifically: how do you see that it does work? [0] Either actively annoyed by it, or learn to ignore it, and hence later on ignore a warning / error you should have read. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM gateways - which one ?
hakem Ta schrieb: I recommand what I know and I sell. Vierling Ecotel VoIP for small capacities and Vierling VTM-pro for ISDN interfaces When I gave the Vierling Ecotel SIP-GSM gateway a try (years ago) it was a nightmare. There was an IP address printed on a sticker on the device - didn't work. And the Ecotel didn't ask my DHCP server for an address either. I used nmap to scan my net for the device. Found it but: no telnet, no web GUI, nothing. When my hair started to turn gray I found out that it came with a MS-Windows-only configuration tool. Great! :-( And - guess what - that tool managed to break almost every human interface guideline. However when you finally have the thing up and running it works. And maybe it's all better now. Philipp Kempgen -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Error Message
Barton Fisher wrote: any ideas? None so far, what version of Asterisk? 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Which non-english language do you have in mind ? Both should differ on this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and ChanSpy strangeness...
Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? Cheers Geraint ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk work with a dynamic IP?
Ronald Wiplinger (Lists) wrote: I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? I had my * server behind my adsl router that was getting a dynamic Ip address. I simply created a domain for my site at http://www.dyndns.com/ (free) and it worked fine. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote: Everyone-- I've just made some major changes to the CDRfix2.rfc.txt file in http://svn.digium.com/svn/asterisk/team/murf/RFCs to accommodate the Leg approach instead of a channel-based approach. Hi murf, I've got a couple of points (as always) from the new design. First one would be the generation of CDRs when putting a call on hold. I don't think that should occur. When a call is put on hold Asterisk never changes the endpoints of a call all it does is possibly change the media to one or both of the call ends. CDRs are about call endpoints not about media transitions. In SIP terms putting a call on hold is no different to changing codecs both operations are re-INVITES and are irrelevant as far as CDRs and billing go. As far as internal calls vs external calls go I would argue that Asterisk can distinguish between them. Any call initiated with the Dial (or equivalent) app is an outgoing call. Anything call request arriving at Asterisk from the outside World is an internal call. For a standard call from a SIP user there are two call legs; the incoming call leg to Asterisk and the outgoing call from the dialplan. For a DAHDI user there is only a single call leg being the outgoing call from the dialplan since providing dialtone when they pick up the phone is not a call leg. I guess it's not really relevant for the CDR design but it's actually not a difficult thing to cope with when writing a billing engine for Asterisk, I know as I've done it. I like the new CDR fields. My number one concern would be to get the CDRs accurate but additional useful information can only help as long as it used the right way, i.e. not treated as definitive for billing purposes. For the linkedid and ideally the uniqueid I reaally think it would be vastly more useful to use a GUID or UUID rather than a incrementing sort of unique id. A lot of use are dealing with CDRs by writing them to databases and it would greatly simplify and improve the robustness of billing if Asterisk CDRs could be categorically be indentified as unique. If we need to know which CDR came first we can use the calldate ther is no need for the linkedid or uniqueid to double up for that. I'm not to sure about: In a leg-based sort of system, CDRs would follow bridging. Does that mean whenever the end of a bridge changes a CDR is generated? And does it mean there are two CDRs per bridge or one? From your examples there only seems to be one CDR per bridge which straight away I can think of a scenario that would cause a problem. If I supply toll free numbers that need to be billed for incoming calls and that can be forwarded out to billable destinations then I want a CDR for both ends of the bridge. In your first blind transfer example what if the initial incoming call to A is billable? I can't see any easy way to get the duration of that call leg. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which installation policy is behind Asterisk doc delivered with source code ?
2008/12/2 Tzafrir Cohen [EMAIL PROTECTED] On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote: Hi, Testing latest 1.6.1, it occurred to me I had to add a couple of noload statements in /etc/asterisk/modules.conf to remove ERROR messages, when starting Asterisk. (I don't imply those ERROR messages were fatal to Asterisk but as a general rule, I tried to start Asterisk without any of those). Could you be specific? I think that some messages may be wrongly-labeled. From memory, I had for instance Failed to open /dev/dahdi/transcode: No such file or directory on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with check hardware compliance. So maybe, the policy is to have a somehow functional system with any of mentioned compliant hardware, with all enhancements such as running as non-root, set apart. As I'm not very familiar with module concepts at the moment, I told myself it would be helpful, if strictly following instructions included in README files, I did get a system that starts without any ERROR message of any kind and still could provide some basic telephony services. So my question is : - is there a policy fixing the target of README files ? - for example, if someone installs Asterisk (according README files) on a platform equipped with a Digium analog board, should this board be automatically discovered, configured and ready to run ? Not yet: # which modules to load: a temporary workaround That's the point : if optimizing modules load could be postponed to a later stage, that would be better, IMHO, as installing from source is already a long process. # dahdi_modules is a simple two-liner scrippt that is currently not # installed by default. I figure I should get that functionality added # to dahdi_genconf dahdi_modules /etc/dahdi/modules /etc/init.d/dahdi start # generate /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf dahdi_genconf dahdi_cfg # edit chan_dahdi.conf accordingly. e.g.: echo '#include dahdi-channels.conf' /etc/asterisk/chan_dahdi.conf # apply changes: start/restart asterisk, or: asterisk -rx 'dahdi restart' IMHO, setting a policy would help to guide efforts of many people involved. That done, maybe we would conclude an interactive script would be the missing piece to incorporate user choices that are hard to default to. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.22 crashing on Solaris in ast_dynamic_str_thread_build_va
Hello, Asterisk 1.4.22 keeps crashing on Solaris 5.10 i386. ast_dynamic_str_thread_build_va() seems to be passed some kind of garbage (see attached dbx output) which ultimately brings down the whole process. As a workaround, I've set the debug level to 0 for now. Should I submit this as a bug? Thanks for any help. Best, Peter [EMAIL PROTECTED] ([EMAIL PROTECTED]) terminated by signal SEGV (no mapping at the fault address) 0xfed1587c: strlen+0x000c: movl (%eax),%edx Current function is ast_dynamic_str_thread_build_va 1354 res = vsnprintf((*buf)-str + offset, (*buf)-len - offset, fmt, ap); (dbx) where current thread: [EMAIL PROTECTED] [1] strlen(0x0), at 0xfed1587c [2] _ndoprnt(0xfe8eb5aa, 0xfc5188e4, 0xfc518130, 0x0), at 0xfed6db66 [3] vsnprintf(0x81fdbcc, 0xb8, 0xfe8eb55c, 0xfc5188e4, 0x81542b0, 0xfedbf000), at 0xfed70c9b =[4] ast_dynamic_str_thread_build_va(buf = 0xfc518178, max_len = 1024U, ts = 0x814a9a0, append = 0, fmt = 0xfe8eb55c Feature interpret: chan=%s, peer=%s, code=%s, sense=%d, features=%d dynamic=%s\n, ap = 0xfc5188e4 çÃ$^H^?Ã$^HÃ\x8aQü^A), line 1354 in utils.c [5] ast_log(level = 0, file = 0xfe8ea4cd res_features.c, line = 1147, function = 0xfe8ea2ab ast_feature_interpret, fmt = 0xfe8eb55c Feature interpret: chan=%s, peer=%s, code=%s, sense=%d, features=%d dynamic=%s\n, ...), line 807 in logger.c [6] ast_feature_interpret(chan = 0x827bc10, peer = 0x826a3b0, config = 0xfc518d50, code = 0xfc518ac0 1, sense = 1), line 1147 in res_features.c [7] ast_bridge_call(chan = 0x827bc10, peer = 0x826a3b0, config = 0xfc518d50), line 1626 in res_features.c [8] dial_exec_full(chan = 0x827bc10, data = 0xfc51bbe0, peerflags = 0xfc519af4, continue_exec = (nil)), line 1780 in app_dial.c [9] dial_exec(chan = (nil), data = (nil)), line 1834 in app_dial.c [10] pbx_extension_helper(c = (nil), con = 0xfc51de18, context = 0x827bd90 outbound_nextra, exten = 0x827bde0 421912345678, priority = 7, label = (nil), callerid = 0x81751f8 421212345678, action = E_SPAWN), line 35 in strings.h [11] __ast_pbx_run(c = (nil)), line 2317 in pbx.c [12] pbx_thread(data = (nil)), line 2621 in pbx.c [13] dummy_start(data = (nil)), line 912 in utils.c [14] _thr_setup(0xfec6ba00), at 0xfed944c7 [15] _lwp_start(0x0, 0xb8, 0xfc5181bc, 0xfedbf000, 0xfc518114, 0x0), at 0xfed947b0 (dbx) threads [EMAIL PROTECTED] a [EMAIL PROTECTED] ?() LWP suspended in __pollsys() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __pollsys() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x8150a20 in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __pollsys() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x818d7cc in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x818e90c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x818fa4c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x8190b8c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x8191ccc in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x8192e0c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x81da874 in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x81db95c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x81dca44 in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x81ddb2c in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __lwp_unpark() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __pollsys() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in ___nanosleep() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0xfd925420 in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() sleep on 0x8150840 in __lwp_park() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __pollsys() [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() LWP suspended in __pollsys() o [EMAIL PROTECTED] a [EMAIL PROTECTED] dummy_start() signal SIGSEGV in strlen() (dbx) thread -info [EMAIL PROTECTED] Thread [EMAIL PROTECTED] (0xfec6ba00) at priority 0 state: active on [EMAIL PROTECTED] base function: 0x80f03b4: dummy_start() stack: 0xfc51e000[245760] flags: DETACHED|SUSPENDED masked signals: HUP INT PIPE TERM WINCH Currently active in strlen ___ --
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
For a ptmp setup where you have multiple phones. Or even a single phone if the port is set to ptmp. Proof of this point is the way I am using our B410P card. Ports 1 and 2 are TE (ptp) and ports 3 4 are NT (ptmp). I have a single ISDN modem connected to port 3 and the B410P would not even look at it unless the 100ohm termination was switched on. So, to reiterate - ptp needs no 100ohm termination (because the end point provides it - aka TEI 0, but ptmp does - aka TEI 127). Looks like we are going to agree to disagree on this one. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another - Email found in subject
On Tue, Dec 02, 2008 at 10:02:06AM -, Andrew Thomas wrote: asterisk-users@lists.digium.com has now been added to the filters white list! Anyway, 100ohm termination isn't required for ptp - but is required for ptmp. For a ptmp setup where you have multiple phones. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another - Email found in subject
asterisk-users@lists.digium.com has now been added to the filters white list! Anyway, 100ohm termination isn't required for ptp - but is required for ptmp. I know the DAHDI package(s) no longer include make b410p - hence the reason it is included in the docs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Dial M option from extensions.ael
Philipp Kempgen schrieb:
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
I tried both (and variations of those with ^ instead of ,) :
Dial(Local/${EXTEN:1},,M(mymacro(${EXTEN}));
Dial(Local/${EXTEN:1},,M(mymacro,${EXTEN}));
both lead to unexpected results:
macro_exec: No such context 'macro-mymacro' for macro 'mymacro'
I'd say use the U option instead of G.
... instead of M.
AEL macros are converted to Gosub routines, not Macros.
Philipp Kempgen
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Re: [asterisk-users] Which installation policy is behind Asterisk doc delivered with source code ?
On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote: Hi, Testing latest 1.6.1, it occurred to me I had to add a couple of noload statements in /etc/asterisk/modules.conf to remove ERROR messages, when starting Asterisk. (I don't imply those ERROR messages were fatal to Asterisk but as a general rule, I tried to start Asterisk without any of those). Could you be specific? I think that some messages may be wrongly-labeled. As I'm not very familiar with module concepts at the moment, I told myself it would be helpful, if strictly following instructions included in README files, I did get a system that starts without any ERROR message of any kind and still could provide some basic telephony services. So my question is : - is there a policy fixing the target of README files ? - for example, if someone installs Asterisk (according README files) on a platform equipped with a Digium analog board, should this board be automatically discovered, configured and ready to run ? Not yet: # which modules to load: a temporary workaround # dahdi_modules is a simple two-liner scrippt that is currently not # installed by default. I figure I should get that functionality added # to dahdi_genconf dahdi_modules /etc/dahdi/modules /etc/init.d/dahdi start # generate /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf dahdi_genconf dahdi_cfg # edit chan_dahdi.conf accordingly. e.g.: echo '#include dahdi-channels.conf' /etc/asterisk/chan_dahdi.conf # apply changes: start/restart asterisk, or: asterisk -rx 'dahdi restart' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. But above, you can read [general] t38pt_udptl = yes Has this parameter name changed between 1.4 to 1.6 from t38_udptl to t38pt_udptl ? A asterisk remains silent when I add an unknown parameter foo=bar, it would perfect if someone could point the right name (t38_udptl or t38pt_udptl). Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Dial M option from extensions.ael [SOLVED]
2008/12/2 Philipp Kempgen [EMAIL PROTECTED]
Philipp Kempgen schrieb:
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party
answers).
It seems to me that asterisk keeps looking for this macro in
extensions.conf
and not in extensions.ael.
I tried both (and variations of those with ^ instead of ,) :
Dial(Local/${EXTEN:1},,M(mymacro(${EXTEN}));
Dial(Local/${EXTEN:1},,M(mymacro,${EXTEN}));
both lead to unexpected results:
macro_exec: No such context 'macro-mymacro' for macro 'mymacro'
I'd say use the U option instead of G.
... instead of M.
AEL macros are converted to Gosub routines, not Macros.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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It works !
I'll take time to edit voip-info.org accordingly ...
Thanks !!
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Re: [asterisk-users] Is HPEC compliant with B410P ?
Olivier wrote: As latest asterisk-libpri-dahdi is introducing dahdi support of B410P, can we use High Performance Echo Canceling addon with B410P ?* Yes, DAHDI echo cancellers work with any DAHDI supported interface. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc and hash problem
On Tuesday 02 December 2008 01:21:46 Giedrius Augys wrote:
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}
extensions.conf
exten = s,1,Ringing
exten = s,n,Wait(4)
exten = s,n,Answer
exten = s,n,Set(NUMERIS=37037210602)
exten = s,n,Set(HASH(RESULTATAS)=${ODBC_GETNUMBER(${NUMERIS})})
exten = s,n,Verbose(1, Number is ${HASH(RESULTATAS, number)}.)
exten = s,n,Verbose(1, Realus 1 ${HASH(RESULTATAS, real_number1)}.)
exten = s,n,Verbose(1, Realus 2 ${HASH(RESULTATAS, real_number2)}.)
exten = s,n,Verbose(1, Statusas ${HASH(RESULTATAS, status)}.)
Kill the space after the comma. You're looking for fields whose names are
number and status, which, of course, don't exist.
--
Tilghman
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Re: [asterisk-users] cepstral vs festival
In my experience cepstral has always had much nicer sounding voices, but I haven't tinkered too much with either. There is a reason one is pay and one free though J I believe cepstral is still offering demo's, I'd download each and see which one gives you the performance you're looking for. Thanks, Matt G : http://www.voipphreak.ca http://www.voipphreak.ca : http://www.ratemydialplan.com http://www.ratemydialplan.com : http://www.asterisk-jobs.com http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Festival is a free voice that sounds like a machine. Cepstral is a fee based human voice ($30 USD per voice per CPU). They are similar in that they both produce mechanically timed output. IMO, you should use festival if this isn't a customer based interface. If it is a CBI, use cepstral and if you don't like it, recreate the wav files it plays (The English language is only based on about 1700 sounds). Cepstral is your choice if your IVR is going to be asterisk interlaced since all asterisk voices are Cepstral Allison out of the can. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help for transfer
Hi All, I need to stop the transfer feature on particular sip user. I am using linksys phone and it has set the forwarding enable to another user. I have three users 2101, 2102, 2103. 2102 is registered in linksys phone with forwarding enable to 2103. But is there any procedure in asterisk that we can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved Temporarily. please advice me that how can we set the user for not to forward or transfer on 2103. i have tested with allowtransfer=no in sip. Thanks in advance! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Desgin
On Mon, Dec 1, 2008 at 7:57 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: JD schrieb: As to the idea of piping to a deamon via socket or dbus: how would asterisk behave if the daemon froze or worse, it lagged? I have implemented something similar with the Dial command. We had a customer that required real-time call control where every 60s a new request had to be made to reserve quota to allow the call to continue. Part of that involved sending a Curl request from app_dial.c as soon as the call answered and as soon as the call hungup. It works well and is loosely coupled with Asterisk so that in the worst case the Curl request times out after 2s and Asterisk could take alternative action if necessary. A nice side effect of this approach is that it's possible to get a realtime display of calls in progress from the database rather than having to use MAPI. Ultimatley I think CEL is going to be the approach for tracking call flow. Hopefully the CEL hooks will be two way so I don't have to keep updating my real-time call control patch for each version of Asterisk :-(. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Festival sucks. Cepstral sucks less. The End. In my experience, it depends on the specific app, who's paying, and who's going to be the victim, err...user listening to it. This is the difference between domain/context specific phrases/words to pronounce vs. general stuff, a client on a tight budget or not, the users being internal vs. customers/public, and so on. Cepstral is a $30 TTS engine. It's not too bad, but you'll find mostly things like Realspeak deployed in large scale professional deployments, such as those used by the big boys, telcos/banks/airlines. We deployed Cepstral recently for a client, for a phone-in service used by the general public, and I can tell you that there was quite a bit of work in teaching it with SSML how to pronounce stuff. Again, it really depends on your specific situation. You should definitely try out those two at least and also ensure that the client/stakeholders are aware of limitations. There's a certain expectation of it will speak perfectly these days, followed by disappointment and blame when reality hits them. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Eric Fort [EMAIL PROTECTED] Sent: Tuesday, December 02, 2008 3:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Log file warnings from chan_sip in build_reply_digest
Using Asterisk 1.4.21.2 I am seeing pairs of warning logs of the form: asterisk[1432]: WARNING[1432]: chan_sip.c:11629 in build_reply_digest: use realm [x] from peer [x][x] These occur once an hour and the x matches the account name for my ITSP. My sip.conf setup for this account is a block copy from an older one that gave no warnings on my old Asterisk v1.2 setup: [general] register = x:[EMAIL PROTECTED] [authentication] auth = x:[EMAIL PROTECTED] [itsp] context=default type=friend username=x user=x host=proxy.itsp.net secret=secret insecure=very accountcode=ITSP amaflags=billing Calls in and out work fine, but I would like to to have a warning free error log if possible. It looks to me from http://www.asterisk.org/doxygen/1.4/chan__sip_8c.html#12860c93a5a453831f5cf15f747fb9fa that this message might be generated anytime you are dealing with an authenticated peer. If so, then this might be a normal but very annoying message. But on the chance that an error on my configuration, I'd like to squelch it. Any suggestions? Thanks! smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SPAM] - MySQL Error Message - Email found in subject
Give this a go:
exten = s,n,MYSQL(Query resultid ${connid} SELECT `name` FROM `cnam`
WHERE `ani` = '${CALLERID(number)}')
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Re: [asterisk-users] cepstral vs festival
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? On Tue, 2 Dec 2008, Matt Gibson wrote: In my experience cepstral has always had much nicer sounding voices, but I haven't tinkered too much with either. There is a reason one is pay and one free though J I believe cepstral is still offering demo's, I'd download each and see which one gives you the performance you're looking for. Way back in the day, festival was awful and Cepstral as almost acceptable. Now, especially with their Allison font, Cepstral is good enough than you can't always tell the difference -- even without using their markup language. The fit with the live Allison's prompts included with Asterisk is great. It's fantastic for demos. You can refine the wording of your prompts before committing to live talent. You may decide that the tts prompts are good enough. I invoke swift (Cepstral's command line tts tool) to create my prompts from my makefile so it's easy to make changes and everything is documented. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
On Dec 2, 2008, at 7:01 AM, Grey Man wrote: On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote: Everyone-- I've just made some major changes to the CDRfix2.rfc.txt file in http://svn.digium.com/svn/asterisk/team/murf/RFCs to accommodate the Leg approach instead of a channel-based approach. Hi murf, I've got a couple of points (as always) from the new design. First one would be the generation of CDRs when putting a call on hold. I don't think that should occur. When a call is put on hold Asterisk never changes the endpoints of a call all it does is possibly change the media to one or both of the call ends. CDRs are about call endpoints not about media transitions. In SIP terms putting a call on hold is no different to changing codecs both operations are re-INVITES and are irrelevant as far as CDRs and billing go. While I agree with your reasoning, I really like the idea of the CDR showing HOLD states. It allows me to generate a report on how often people are on hold. If I see that the incoming calls to my receptionist spend 15% of the time on hold, that means something to me. If someone doesn't care to know the hold states, they (or their script) can just ignore the HOLD CDR records. I don't see that it would impact any final numbers to just skip them, you still get the total call duration between point A and point B. Daniel Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging, Polycom and whispers
Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging, Polycom and whispers
Mike wrote: Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? I've never been able to find a way. Any attempt I made either put the existing call on hold to auto-answer the page or the page just rang at the phone and then caused other issues. I'm not sure you'll have any luck with other SIP phones either. What you're asking it to do is accept two simultaneous calls but put each call on a different listening device (handset/speakerphone in this case). The closest you might get is to rig a dialplan that would use chanspy in whisper mode to play the page through the current audio device if the phone is busy. I don't know how to go about doing that however. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc and hash problem
2008/12/2 Tilghman Lesher [EMAIL PROTECTED]
On Tuesday 02 December 2008 01:21:46 Giedrius Augys wrote:
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM
ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}
extensions.conf
exten = s,1,Ringing
exten = s,n,Wait(4)
exten = s,n,Answer
exten = s,n,Set(NUMERIS=37037210602)
exten = s,n,Set(HASH(RESULTATAS)=${ODBC_GETNUMBER(${NUMERIS})})
exten = s,n,Verbose(1, Number is ${HASH(RESULTATAS, number)}.)
exten = s,n,Verbose(1, Realus 1 ${HASH(RESULTATAS, real_number1)}.)
exten = s,n,Verbose(1, Realus 2 ${HASH(RESULTATAS, real_number2)}.)
exten = s,n,Verbose(1, Statusas ${HASH(RESULTATAS, status)}.)
Kill the space after the comma. You're looking for fields whose names are
number and status, which, of course, don't exist.
--
Tilghman
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But I noticed, that using isql utility I also don't see column name...
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Paging, Polycom and whispers
You can send an IM to the phone with a text message. Assuming that the phone has more than 1 line and at least one is open, the call should go through without effecting the existing call. To do this from the dialplan, you could set up something like this: Exten = 411,1,Dial(SIP/100,1) Exten = Sendtext(You have a call on park 701) Exten - hangup(} This also assumes that the polycom has presence enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, December 02, 2008 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging, Polycom and whispers Mike wrote: Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? I've never been able to find a way. Any attempt I made either put the existing call on hold to auto-answer the page or the page just rang at the phone and then caused other issues. I'm not sure you'll have any luck with other SIP phones either. What you're asking it to do is accept two simultaneous calls but put each call on a different listening device (handset/speakerphone in this case). The closest you might get is to rig a dialplan that would use chanspy in whisper mode to play the page through the current audio device if the phone is busy. I don't know how to go about doing that however. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Bridge Application
Hi, i am running Asterisk 1.6.0-beta4 and i have some trouble with the Bridge-Application. Here is what i want to do: 1) Caller A calls an extension and is connected to an AGI-Script. 2) Doing stuff and originating a second call per Manager Interface 3) Call will be set to an extension with MusicOnHold 4) Caller A hears MusicOnHold 5) Meanwhile, the second call is established and is also connected to an AGI Script 6) Doing stuff 7) Since we have transported the channel name of the first call to the agi script we can execute the Bridge Application in order to bridge the two channels. After executing the Bridge-Application MusicOnHold is stopped on the first call, but the second call in HungUp immediatly. On the Asterisk CLI i see that the correct channel names are issued. There seems to an issue with the second call, which is originated by the asterisk server. If i dial the extension manually, i get connected to the agi which executes Bridge(correct channel name) and everything works fine. Any ideas? Regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Bridge Application
Right after sending the email, the solution came to me. I have fooled myself: A ManagerEventListener kicked in an issued an HangUp Action on the second channel right after the Bridge ... The Bridging workes perfectly after fixing the EventListener. Have a nice day, i will go home and hit myself a little bit ... Tobias Wolf schrieb: Hi, i am running Asterisk 1.6.0-beta4 and i have some trouble with the Bridge-Application. Here is what i want to do: 1) Caller A calls an extension and is connected to an AGI-Script. 2) Doing stuff and originating a second call per Manager Interface 3) Call will be set to an extension with MusicOnHold 4) Caller A hears MusicOnHold 5) Meanwhile, the second call is established and is also connected to an AGI Script 6) Doing stuff 7) Since we have transported the channel name of the first call to the agi script we can execute the Bridge Application in order to bridge the two channels. After executing the Bridge-Application MusicOnHold is stopped on the first call, but the second call in HungUp immediatly. On the Asterisk CLI i see that the correct channel names are issued. There seems to an issue with the second call, which is originated by the asterisk server. If i dial the extension manually, i get connected to the agi which executes Bridge(correct channel name) and everything works fine. Any ideas? Regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tue, Dec 2, 2008 at 8:22 PM, Dave Fullerton [EMAIL PROTECTED] wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ACK [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 On what platform is it? I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
Tzafrir Cohen wrote: On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 On what platform is it? Slackware 12.0 1.6.0.1 compiles fine. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
It bombs out when compiling manager.c On what platform is it? Fails on CentOS 5x86 as well. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tue, Dec 2, 2008 at 9:56 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 On what platform is it? Fedora Core release 6 (Zod) - Linux ast-dev14 2.6.21.1skvt #1 Fri May 18 10:14:35 EEST 2007 i686 i686 i386 GNU/Linux Fedora release 8 (Werewolf) - Linux asterisk-dev-mc 2.6.24.7-92.fc8 #1 SMP Wed May 7 16:26:02 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Debian Etch (4.0) - Linux saule 2.6.18-6-xen-686 #1 SMP Thu May 8 11:28:36 UTC 2008 i686 GNU/Linux Debian Sid - Linux debian 2.6.26-1-686 #1 SMP Thu Oct 9 15:18:09 UTC 2008 i686 GNU/Linux 1.6.0.1 compiled fine on at least two Fedoras. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hi from argentina
hi this is mi first email and just for say hello. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy strangeness...
Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? I don't have any suggestions, but this is similar to something I am experiencing with Chanspy in 1.4.21.1. If I spy on a call, then progressively throughout the call a delay is introduced. By the end of the call I can be listening to sound that is 10 seconds out of sync. (Then I don't get to hear the end of the call when the call is finished). This also leaves stale channels open. (the entry in show channels doesn't go away until the asterisk process is restarted). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, December 02, 2008 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2,1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released On Tue, Dec 2, 2008 at 8:22 PM, Dave Fullerton [EMAIL PROTECTED] wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ACK [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Regards, Atis Looks like branches/1.6 got the trunk version of a fix to OpenBSD compilation rather than the 1.4 version as it should have. 1.4: http://svn.digium.com/view/asterisk/branches/1.4/main/manager.c?view=dif frev=159897r1=159896r2=159897 Trunk: http://svn.digium.com/view/asterisk/trunk/main/manager.c?view=diffrev=1 59898r1=159897r2=159898 - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
At 04:03 12/2/2008, Benny Amorsen wrote: Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. What happens if everyone who owns a car drives it at the same time? Owns a telephone and uses it at the same time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridging - Didn't get a frame from channel
Hi, I appear to have fixed this issue. If I turn off call recording in queues.conf, the bridging succeeds. I did this by commenting monitor-format and monitor-type. thanks, Tony Gaspar From: Tony Gaspar [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 26 November, 2008 12:14:31 PM Subject: [asterisk-users] bridging - Didn't get a frame from channel Hi, I am having a difficulty with getting two realtime user’s to bridge on answer. I have managed successfully to bridge the same two users/channels via the Bridge Manager api command and confirm that the two communicate directly bypassing the asterisk server (I confirmed this with Wireshark). Does anyone have some ideas? I have put some log entries below. I haven’t attached my dialplan. Some behaviours I have discovered are: 1) If user A dials directly to the call number of user B, bridging works on answer and the failed log entries below don’t occur. 2) If user A dials into a queue and waits for User B (who is registered in the queue). Then when User B answers, the bridging failures occur as in the log entries below. Thank you. Tony Gaspar [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=26) [Nov 25 17:47:38] DEBUG[27462] devicestate.c: Notification of state change to be queued on device/channel IAX2/asxop [Nov 25 17:47:38] VERBOSE[27462] logger.c: -- IAX2/asxop-14202 answered IAX2/wally-10884 [Nov 25 17:47:38] DEBUG[25220] devicestate.c: No provider found, checking channel drivers for IAX2 - asxop [Nov 25 17:47:38] DEBUG[25220] chan_iax2.c: Checking device state for device asxop [Nov 25 17:47:38] DEBUG[25220] chan_iax2.c: iax2_devicestate: Found peer. What's device state of asxop? addr=984047307, defaddr=0 maxms=5000, lastms=191 [Nov 25 17:47:38] DEBUG[25220] devicestate.c: Changing state for IAX2/asxop - state 2 (In use) [Nov 25 17:47:38] DEBUG[25225] app_queue.c: Device 'IAX2/asxop' changed to state '2' (In use) [Nov 25 17:47:38] DEBUG[27462] app_queue.c: Next is 'IAX2/mark' with metric 1 [Nov 25 17:47:38] DEBUG[27462] channel.c: Set channel IAX2/wally-10884 to write format ilbc [Nov 25 17:47:38] VERBOSE[27462] logger.c: -- Stopped music on hold on IAX2/wally-10884 [Nov 25 17:47:38] DEBUG[27462] channel.c: Scheduling timer at 0 sample intervals [Nov 25 17:47:38] DEBUG[27462] app_queue.c: Starting MixMonitor as requested. [Nov 25 17:47:38] DEBUG[27462] app_queue.c: Arguments being passed to MixMonitor: 1227656841.388.wav,b [Nov 25 17:47:38] DEBUG[27462] app_queue.c: Queue 'assist' Leave,
Re: [asterisk-users] OT: What do you guys think of this?
At 07:00 12/2/2008, SIP wrote: Doug wrote: At 18:56 12/1/2008, Tilghman Lesher wrote: On Monday 01 December 2008 06:21:33 pm Doug wrote: We tell our customers that they are not allowed to download copyrighted material. So your customers are only allowed to download public domain material? That kind of restricts the amount of information available on the Internet. Nitpick: just about everything, including this email, is copyrighted by somebody. Forbidding the download of copyrighted works is not only a draconian policy, but may actually violate several copyright laws (you're interfering with a copyright owner's right to distribute his/her/their works, and courts are generally not very sympathetic with your position). Oops! Didn't mean to start a fire here. I meant to say illegal copyrighted material. Also, if they are using up hundreds of Internet connections, we can see that. It essentially causes a Denial of Service situation for other users on that leg of our wireless network. The system supposedly has rate limiting, but seems to get overloaded when someone goes completely nuts with BitTorrent. We are working on ways to limit the number of simultaneous connections. When we get a copyright infringment notice from our upstream provider, we are compelled to reprimand the user. I don't think we have sent a customer to the shower even if they had several notices. Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Es tut mir leid für das Durcheinander meine Brüder! This is the classic logical fallacy that people seem to perpetuate when reporting news about P2P activity. ISPs oversubscribe. It's a common practice, and reasonably valid. But when you oversubscribe, you use a model based on 'projected' use of the available circuits and bandwidth. If you have a user who pays for a circuit that you've advertised as an X Mb line, and he uses X Mb ALL the time, he's using what he's paying for. If you then proceed to tell him that he can't do that, you're either wrong or you're not being up front enough with your pricing and marketing materials. You can't then proceed to blame the customer for use you did not anticipate. Imagine a farmer who sells tomatoes. He's promised you a bushel, but he gets a harvest of only so many. You walk up to the counter just after he's sold all of his tomatoes to someone and he tells you Sorry. There are no more tomatoes because that customer before you just 'stole' them all from you. He's abusing his privileges by buying up my whole crop. Now whose fault is it that you don't get the tomatoes you want? Is it the customer's fault for buying all the tomatoes the farmer sold him? Or is it the farmer's fault for selling them? The same works with the ISP vs P2P argument. If the ISPs were up-front about saying that they do not intend for you to actually USE the bandwidth you think you're paying for, I would say they had a leg upon which to stand. However, hiding this information from the customer and then blaming the customer when he does what he believes is well within his rights... it may play well in the media, but it's bad for the whole system and is incredibly divisive. Yep. In our contract we say things like shared, best efforts, etc. If you want a dedicated pipe with guaranteed bandwidth, you gotta pay a hefty price. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 04:03 12/2/2008, Benny Amorsen wrote: Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. What happens if everyone who owns a car drives it at the same time? Owns a telephone and uses it at the same time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If everyone who owns a car drives it at the same time, there's lots of traffic. You know who gets blamed? The right people -- the people to create the infrastructure. Drivers aren't blamed for driving their cars when they want to as long as they do it legally as prescribed by the very open and easy to find laws. If everyone who owns a telephone uses it at the same time, it's just like the Internet issues. Telephone companies also practice oversubscription. But it's clear to everyone that it's the phone company that doesn't have the capacity for it... people don't blame the customers for using their phone. They pay for it. They should be able to use it when they want. But if everyone uses the Internet access they pay for? Suddenly, they're violating a user agreement (usually not a specified one in the case of many ISPs) or a usage policy and it's all that crazy P2P to blame. They're stealing bandwidth from other users. Which is absolute poppycock. That's a marketing spin on poor infrastructure planning. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
At 07:57 12/2/2008, Andrew Kohlsmith (lists) wrote: On December 1, 2008 07:21:33 pm Doug wrote: Hmmm. When our users are pounding the network with BitTorrent traffic, we just shut them down and wait for them to complain. It's against our Acceptable Use Policy, and causes all sorts of VOIP headaches. As someone who is the technical lead for several ISPs, it is my professional opinion that you haven't a clue how to run such a thing. Torrent does not interfere with VOIP on a well-designed network any more than FTP or web browsing. Honestly, hire a competent admin to set up and run your infrastructure. If we could find one. We had to completely abandon our initial supplier of wireless point-to-point gear. We are still ramping up with the new vendor. Lots of problems. We keep asking questions--sometimes we get satisfactory answers. This is what life is like on cutting edge of tecnology. If torrent's killing VOIP, that means that adding more VOIP will also kill it. Or excessive web browsing. Thank God I'm not one of your customers. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 07:00 12/2/2008, SIP wrote: Doug wrote: At 18:56 12/1/2008, Tilghman Lesher wrote: On Monday 01 December 2008 06:21:33 pm Doug wrote: We tell our customers that they are not allowed to download copyrighted material. So your customers are only allowed to download public domain material? That kind of restricts the amount of information available on the Internet. Nitpick: just about everything, including this email, is copyrighted by somebody. Forbidding the download of copyrighted works is not only a draconian policy, but may actually violate several copyright laws (you're interfering with a copyright owner's right to distribute his/her/their works, and courts are generally not very sympathetic with your position). Oops! Didn't mean to start a fire here. I meant to say illegal copyrighted material. Also, if they are using up hundreds of Internet connections, we can see that. It essentially causes a Denial of Service situation for other users on that leg of our wireless network. The system supposedly has rate limiting, but seems to get overloaded when someone goes completely nuts with BitTorrent. We are working on ways to limit the number of simultaneous connections. When we get a copyright infringment notice from our upstream provider, we are compelled to reprimand the user. I don't think we have sent a customer to the shower even if they had several notices. Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Es tut mir leid für das Durcheinander meine Brüder! This is the classic logical fallacy that people seem to perpetuate when reporting news about P2P activity. ISPs oversubscribe. It's a common practice, and reasonably valid. But when you oversubscribe, you use a model based on 'projected' use of the available circuits and bandwidth. If you have a user who pays for a circuit that you've advertised as an X Mb line, and he uses X Mb ALL the time, he's using what he's paying for. If you then proceed to tell him that he can't do that, you're either wrong or you're not being up front enough with your pricing and marketing materials. You can't then proceed to blame the customer for use you did not anticipate. Imagine a farmer who sells tomatoes. He's promised you a bushel, but he gets a harvest of only so many. You walk up to the counter just after he's sold all of his tomatoes to someone and he tells you Sorry. There are no more tomatoes because that customer before you just 'stole' them all from you. He's abusing his privileges by buying up my whole crop. Now whose fault is it that you don't get the tomatoes you want? Is it the customer's fault for buying all the tomatoes the farmer sold him? Or is it the farmer's fault for selling them? The same works with the ISP vs P2P argument. If the ISPs were up-front about saying that they do not intend for you to actually USE the bandwidth you think you're paying for, I would say they had a leg upon which to stand. However, hiding this information from the customer and then blaming the customer when he does what he believes is well within his rights... it may play well in the media, but it's bad for the whole system and is incredibly divisive. Yep. In our contract we say things like shared, best efforts, etc. If you want a dedicated pipe with guaranteed bandwidth, you gotta pay a hefty price. Then I applaud you for doing something most ISPs do not do -- being a LITTLE more up-front about the realistic limitations of the service. ISPs tend to promise the world to grab users, knowing full well they can't deliver. And when the users try and use what they've been promised, they're blamed for bringing down the network. And what's worse, this clear spin line is propagated throughout even LARGE media organisations as an accepted fact. P2P Steals Bandwidth. That's reported as a simple and plain fact when, in reality, you can't steal what you've been allotted by your ISP. If the ISP said we only have the capacity for X users to use their service ALL the time, so users who want to pay basic usage and use little can pay this small sum, or users who want to get unlimited but very throttled and pay this larger sum, it would go a long way toward fostering trust all-round without relying on misinformation and vilifying the users who are using what they think they're paying for. Of course, it would be a marketing nightmare, as the other ISPs would say, But we have UNLIMITED access at much higher speeds -- clearly lying about their capacities for the sake of bamboozling non-tech-savvy customers, and then relying on media organisations to propagate their disingenuous epithets against the P2P crowd. N.
Re: [asterisk-users] Parking calls
It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on….. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking calls
This seems to be an AGI/Music on Hold solution to me. For parking to work, you would have to know which lot you parked the call in and pick it back up when done, assuming that another user did not pick it up and that the caller did not hang up. From the dialplan, you would call an AGI. The AGI would do something like this: print STDOUT EXEC background /var/lib/asterisk/sounds/wait-moment \n system(program2.agi ) exit; program 2 would run while the sound played. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, December 02, 2008 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on .. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tuesday 02 December 2008 12:22:16 Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? I'll get a new release candidate out either this afternoon or tomorrow; I'm currently working on ensuring that 1.6.0.3 will not be a regression from 1.4.23. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi and ztdummy
Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
Mike wrote: Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,** * * *Mike* DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it exactly the same way that you used ztdummy. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callcenter supervisor system
hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
I guess my question is more basic than that: I have two brand new 1.4.22 systems, one with DAHDI apparently running well (dahdi start looks god) running well with Paging, and the other with FATAL errors modules cannot be found and paging not working. I seem to remember installing both Asterisks the same way. Where should I be looking? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Tuesday, December 02, 2008 17:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy Mike wrote: Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,** * * *Mike* DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it exactly the same way that you used ztdummy. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
there is any card? you musnt load any module that is not going to be used. you can get some errores if the card is misconfigured trai dahdi_cfg -vvv you will get some idea of the problem David 2008/12/2 Mike [EMAIL PROTECTED] I guess my question is more basic than that: I have two brand new 1.4.22 systems, one with DAHDI apparently running well (dahdi start looks god) running well with Paging, and the other with FATAL errors modules cannot be found and paging not working. I seem to remember installing both Asterisks the same way. Where should I be looking? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Tuesday, December 02, 2008 17:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy Mike wrote: Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,** * * *Mike* DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it exactly the same way that you used ztdummy. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
On Dec 2, 2008, at 9:41 AM, Erik (Caneris) wrote: Festival sucks. Cepstral sucks less. The End. In my experience, it depends on the specific app, who's paying, and who's going to be the victim, err...user listening to it. This is the difference between domain/context specific phrases/words to pronounce vs. general stuff, a client on a tight budget or not, the users being internal vs. customers/public, and so on. Cepstral is a $30 TTS engine. It's not too bad, but you'll find mostly things like Realspeak deployed in large scale professional deployments, such as those used by the big boys, telcos/banks/ airlines. We deployed Cepstral recently for a client, for a phone-in service used by the general public, and I can tell you that there was quite a bit of work in teaching it with SSML how to pronounce stuff. Again, it really depends on your specific situation. You should definitely try out those two at least and also ensure that the client/stakeholders are aware of limitations. There's a certain expectation of it will speak perfectly these days, followed by disappointment and blame when reality hits them. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com Erik - Have you found RealSpeak to be worth the cost? Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? It's been a while since I did a head-to-head comparison between Cepstral and (anything else) so I did a quick demo of the RealSpeak Host-based telecom app: http://www.nuance.com/realspeak/demo/ (contact data required) and the Cepstral demo: http://www.cepstral.com/demos/ I used the Jill (default - 8khz) for RealSpeak and Allison (default) for the tests, and played back the same phrase: Congratulations. You have successfully installed and executed the Asterisk open source PBX. My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
Although QM is not open-source, it is extremely affordable and high quality for the price. David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
thanks for your answer but i need an opensource i know quemetrics is good but i need an open source. thanks David 2008/12/2 Alex Balashov [EMAIL PROTECTED] Although QM is not open-source, it is extremely affordable and high quality for the price. David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
John Todd wrote: Erik - Have you found RealSpeak to be worth the cost? Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? It's been a while since I did a head-to-head comparison between Cepstral and (anything else) so I did a quick demo of the RealSpeak Host-based telecom app: http://www.nuance.com/realspeak/demo/ (contact data required) and the Cepstral demo: http://www.cepstral.com/demos/ I used the Jill (default - 8khz) for RealSpeak and Allison (default) for the tests, and played back the same phrase: Congratulations. You have successfully installed and executed the Asterisk open source PBX. My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director This may not be a perfectly fair comparison. Looks like you're comparing the RealSpeak 8khz voice to the Cepstral default Allison which is NOT 8khz. If you look on the Cepstral site you'll see Desktop Voices and Telephony Voices. The Cepstral Telephony voices are 8khz, and I suspect their quality is on par with RealSpeak. I recently licensed the Allison-8Khz voice for some of the admin functions on my companies phone systems where I didn't want to record prompts and Flite was too robotic sounding. The Allison-8khz voice is virtually indistinguishable from the pre-recorded Allison prompts, except for maybe some minor differences in inflection. I was thoroughly impressed with the quality though. For the most part it sounds like you've hired Allison to record custom prompts. The Allison Desktop voice is OK, but sounds sort of like Allison is taking through a spinning fan blade. When you're doing TTS comparisons be sure you're comparing apples to apples and not peaches to apricots. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
Do you need free or do you need opensource? What is your budget? Number of agent seats? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Tue, Dec 2, 2008 at 6:18 PM, David fire [EMAIL PROTECTED] wrote: thanks for your answer but i need an opensource i know quemetrics is good but i need an open source. thanks David 2008/12/2 Alex Balashov [EMAIL PROTECTED] Although QM is not open-source, it is extremely affordable and high quality for the price. David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
line 0: Unable to open master device '/dev/dahdi/ctl Well that probably explains it, because there is no such file. But as I am not a linux expert (comfortable linux user at best), I am not sur where to go next. Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 02, 2008 17:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy there is any card? you musnt load any module that is not going to be used. you can get some errores if the card is misconfigured trai dahdi_cfg -vvv you will get some idea of the problem David 2008/12/2 Mike [EMAIL PROTECTED] I guess my question is more basic than that: I have two brand new 1.4.22 systems, one with DAHDI apparently running well (dahdi start looks god) running well with Paging, and the other with FATAL errors modules cannot be found and paging not working. I seem to remember installing both Asterisks the same way. Where should I be looking? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Tuesday, December 02, 2008 17:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy Mike wrote: Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,** * * *Mike* DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it exactly the same way that you used ztdummy. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking calls
I found other solution, I can use cannel local to dial to an extension with m parameter, then I can put Ringing as the first thing to do that will follow processing the next lines of the dialplan, with the m option MOH will sound instead of ringing, and I can do the heavy work there till I finish and do other things with the call. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: martes, 02 de diciembre de 2008 07:37 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Parking calls This seems to be an AGI/Music on Hold solution to me. For parking to work, you would have to know which lot you parked the call in and pick it back up when done, assuming that another user did not pick it up and that the caller did not hang up. From the dialplan, you would call an AGI. The AGI would do something like this: print STDOUT EXEC background /var/lib/asterisk/sounds/wait-moment \n system(program2.agi ) exit; program 2 would run while the sound played. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, December 02, 2008 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on .. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3659 (20081202) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3659 (20081202) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
my budget is 0 rigth now and i want opensource because i want to customice it... can program in Java PHP C C++ .NET (i am not proud of it) so i want to customice it for my clients (furute clients i havent any now) and give the improvements to the comunity. Davif 2008/12/2 Steve Totaro [EMAIL PROTECTED] Do you need free or do you need opensource? What is your budget? Number of agent seats? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Tue, Dec 2, 2008 at 6:18 PM, David fire [EMAIL PROTECTED] wrote: thanks for your answer but i need an opensource i know quemetrics is good but i need an open source. thanks David 2008/12/2 Alex Balashov [EMAIL PROTECTED] Although QM is not open-source, it is extremely affordable and high quality for the price. David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
I have no cards (nothing dahdi related). Why is my other server, built with default settings, working then? Still what do I do ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 02, 2008 19:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
Sorry, that worked, I just removed all modules from the modules file in /etc/dahdi. But that doesn`t explain why my other card-free PC is working perfectly with the default modules files while this one isn`t Thanks though, that saved my behind. But an explanation, if an easy one can be found, would be appreciated. Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 02, 2008 19:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
At 12:44 PM 12/2/2008, you wrote: At 04:03 12/2/2008, Benny Amorsen wrote: Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. What happens if everyone who owns a car drives it at the same time? Owns a telephone and uses it at the same time? As far as I remember the very first service to offer flat rate was BIX. They very carefully figured out what it would cost to insure a fair profit, and it was a big hit till a few people figured out that they could use private chats as a network pipe and stay on 24/7 using some mysterious protocol. In the end, that was some of what killed the service and there was nothing to be done about it. For most of us, well for me anyway, I like the fat pipe I have for the 1% of the time I use it and I expect that as a residential user Time Warner sell me that pipe expecting me to use it about that much, maybe a bit more if I had teenage kids. I'm sure in the fine print it says I can't host a web server though I'd guess they'd not complain if it didn't get much traffic. I've considered a T1 so I'd be guaranteed the throughput so my phones would always work, but that costs 10 times as much and has less promised speed than my cable modem. So personally I consider that if I was to try and use my current internet connection to host a torrent site and it tried to use 100% of the promised capacity all the time that I'd get cut off. The same as most of the unlimited phone service says in fine print up to 2000 minutes/month or some such limit. If I could get the same plan for my internet as I get for my phones, a few dollars a month plus a bit per minute(megabyte), I'd be all over it, but even better, then the provider wouldn't have to care as they'd be making a fair profit no matter what the user did. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
Have you looked at AMS? http://www.intuitivecreations.com/contributions/AMS/ PaulH David fire wrote: my budget is 0 rigth now and i want opensource because i want to customice it... can program in Java PHP C C++ .NET (i am not proud of it) so i want to customice it for my clients (furute clients i havent any now) and give the improvements to the comunity. Davif 2008/12/2 Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Do you need free or do you need opensource? What is your budget? Number of agent seats? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Tue, Dec 2, 2008 at 6:18 PM, David fire [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: thanks for your answer but i need an opensource i know quemetrics is good but i need an open source. thanks David 2008/12/2 Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Although QM is not open-source, it is extremely affordable and high quality for the price. David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
options are: -you made a mistake -only g'ds know probably when you installed dahdi you made make config in only one pc. David 2008/12/2 Mike [EMAIL PROTECTED] Sorry, that worked, I just removed all modules from the modules file in /etc/dahdi. But that doesn`t explain why my other card-free PC is working perfectly with the default modules files while this one isn`t… Thanks though, that saved my behind. But an explanation, if an easy one can be found, would be appreciated. Mike *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *David fire *Sent:* Tuesday, December 02, 2008 19:00 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Dahdi and ztdummy ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM gateways - which one ?
Hello Philipp, All what you says was true and I am first to witness it. Ecotel VoIP was a new product at that time needed some time like wine to become good 1-) Files did not cleanup correctly ; 2-) Lot of erratic issues that we have also greyed hair with; 3-) Stability issues, ...etc; When I gave the Vierling Ecotel SIP-GSM gateway a try (years ago) it was a nightmare. There was an IP address printed on a sticker on the device - didn't work. And the Ecotel didn't ask my DHCP server for an address either. (HT:) Ecotel VOIP allows USB connectivity first. Through USB cable, you can assign to it IP address. New models allow also GSM calls remotely; I used nmap to scan my net for the device. Found it but: no telnet, no web GUI, nothing. (HT): There is Service Gear GUI application that you can download. When my hair started to turn gray I found out that it came with a MS-Windows-only configuration tool. Great! :-( (HT): Indeed. Although it is linux based box, config through GUI is windows based. It handles sim rotation and advanced sim management scenarii; And - guess what - that tool managed to break almost every human interface guideline. (HT): I hope you did not throw your box ! In case not, you can try it and I can help you make it work (albeit today is stable that we use in production); However when you finally have the thing up and running it works. And maybe it's all better now. (HT): Indeed. Despite other products we have tried on the market protech and many other products (voiceblue of 2N), Ecotel VoIP can play VoIP, ISDN, GSM (CDMA ) + sip proxy (SER based). It can work in NAT with very advanced sim rotation capabilities (if need be to use lot of sims for free packages or so). It has advanced logging (you can trace SER and also GSM AT commands). These make this product not in same range oas protech and other ones alike Philipp Kempgen Hakem, -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Hakem Voip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Todd a écrit : My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) For French language, I find the quality of RealSpeak to be very good. Festival was unusable (for French); I tried Cepstral but was deceived. The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... That's a C library. I bought RealSpeak SDK, and developed app_realspeak for Asterisk (1.2, then ported to 1.4). I've been using it since 2005 for my IVR projects, including telcos/banks/airlines :) Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkk10kAACgkQuu7Rv+oOo/gK2ACfXedtJ8k7cmVRpOqTU+rYpbVy PcIAnjbXbDPuicE29673TQY3CritOksQ =vvB7 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter supervisor system
David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one You'll be pushing to find something even close to QueueMetrics' quality available in open source. The closest I'm aware of is Vicidial, though if you only want a call centre statistics package, Vicidial doesn't really meet the requirements since it's focus is on being a predictive dialler. There are a couple of unfinished, unpolished packages that are around that don't even come /close/ to what is available through QueueMetrics. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This has been an interesting discussion about cepstral. My question is why it doesn't appear to be available for 1.6 yet? This thread has piqued my interest in the product but a visit to Digium's website seems to point to it being a product for Asterisk 1.6. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJNdMwCFu3bIiwtTARAqqQAJ9mXLMyUCzI+UCiF3/1j4kuGE32ewCgpS2r 8IwCpap3Q1puuP4LZScVV00= =4Cdn -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Jean-Denis Girard wrote: The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. Only an order of magnitude? They've reduced it a lot then. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
Try just modprobing the module and see what happens. This worked for me when it was zaptel. on Tuesday 12/02/2008 Mike([EMAIL PROTECTED]) wrote I have no cards (nothing dahdi related). Why is my other server, built with default settings, working then? Still what do I do ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 02, 2008 19:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Erik - Have you found RealSpeak to be worth the cost? Actually my last note was probably a bit misleading because in the particular cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't even on the radar. Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a client using Asterisk, Cepstral, and a lot of custom code. It's a free phone-in service that allows folks to get local traffic, weather, news, commuter transit, border crossing wait times, and more. There's obviously quite a bit of domain-specific, dynamic, constantly changing text, so this is certainly an example of pushing it to the max. Just think of all the street names it has the potential to mispronounce. It's a work in progress, but it's very promising. Definitely an example of a lot of hourly $ spent on tuning as you put it. My results: The RealSpeak sample was more clear than the Cepstral. Depends on what you mean by more clear. As Brent Davidson mentions, make sure you're comparing 8khz to 8khz, or similar. If you mean it pronounces things better, then I agree. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... MRCP is the standard for interfacing with ASR and TTS engines (including RealSpeak) in other platforms. Brief Googling reveals a previous flame war on asterisk-dev regarding MRCP. I have no idea if it's implemented in Asterisk now. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
2008/12/3 Steve Underwood [EMAIL PROTECTED] Jean-Denis Girard wrote: The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. Only an order of magnitude? They've reduced it a lot then. :-) 1 order of magnitude = x10 Then, shall we say 500$/simultaneous voice ? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
2008/12/3 Joseph L. Casale [EMAIL PROTECTED] Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) HTH, jlc From someone also discovering how to install and configure dahdi : 1. Is it normal to see : # lsmod Module Size Used by dahdi_dummy 3236 0 Shouldn't it be used by asterisk or is this 0 value meaning something specific ? 2. How can you check dahdi is running ? Here, ps aux | grep dahdi replies grep dahdi. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
Hi
I had tried your sugesntion and added in the appropriate context but got
this error message
Rejected connect attempt from 192.168.254.185, request
'[EMAIL PROTECTED]' does not exist
any ideia?
thanks
coco wrote:
Hello
I asked the same thing some time ago, but nobody answered.
I founded some workaround.
Use this in your dialplan:
exten = _7.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:1})
exten = _7.,n,Pickup(${EXTEN:[EMAIL PROTECTED])
This worked for me.
Cosmin
--- On *Thu, 11/27/08, Bruno Castelo Branco
/[EMAIL PROTECTED]/* wrote:
From: Bruno Castelo Branco [EMAIL PROTECTED]
Subject: Re: [asterisk-users] pick up IAX2 calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thursday, November 27, 2008, 4:59 AM
Somebody know some work around for it?
I still trying to find a solution but nothing seems to work
thanks
Eric ManxPower Wieling wrote:
The problem is that IAX2 does not seem to support call pickup.
Bruno Castelo Branco wrote:
hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1
for all IAX extensions in iax.conf. Didn't works for while.
thanks
Tim Panton wrote:
I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in mixture.
I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8
So if you have Dial(IAX/fredSIP/billzap/mark)
then someone in the same group as fred can pickup with IAX
and someone in the same group as bill can pickup with SIP
etc.
So it's an asterisk thing, not an IAX thing per-se.
Tim.
P.S.
(you could try putting in a dummy 'fred' entry into Dial and iax.conf.)
T.
On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote:
hi
thanks Luis , but doesn't work.
For SIP extensions works well *8, but for IAX a tried *8 and ** + iax
extension and didn't works
Luis Morales wrote:
Try with ** + iax extension
Regards,
Luis Morales
On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Hi
Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2 extensions.
Any idea?
thanks
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Re: [asterisk-users] Dahdi and ztdummy
On Tue, Dec 02, 2008 at 06:47:02PM -0500, Mike wrote: line 0: Unable to open master device '/dev/dahdi/ctl Well that probably explains it, because there is no such file. But as I am not a linux expert (comfortable linux user at best), I am not sur where to go next. This probably means that dahdi is not loaded. ls /proc/dahdi ls /sys/class/dahdi lsmod | grep ^dahdi -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
On Wed, Dec 03, 2008 at 07:43:55AM +0100, Olivier wrote: 2. How can you check dahdi is running ? cat /sys/module/dahdi/version -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
2008/12/3 Tzafrir Cohen [EMAIL PROTECTED] On Wed, Dec 03, 2008 at 07:43:55AM +0100, Olivier wrote: 2. How can you check dahdi is running ? cat /sys/module/dahdi/version Thanks ! -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
