Re: [Asterisk-Users] arrgg! www.voip-info.org down again (or too busy)
On Saturday 16 July 2005 16:22, Johan Nordström wrote: Lists skrev: Yes, it's annoying as hell. A few times I've been close to make a mirror for myself so that I can access it reliably. Not quite the same to go through google. I just have to figure out what it would take set one up... I've read yours and others among with you and I totally agree with you. I've mailed the maintainer of the site last night, here's what he responded: Thanks for the kind words. There are no current mirrors -- wikis are somewhat difficult to mirror. I'm working on new software, which I hope will both improve performance, and make it easier to mirror. Jim Excellent, I'm looking forward to greater flexibility! -- List Manager Network Voice Comunications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Validating a phone number
I'm concerned about people dialing out of our asterisk server to numbers they shouldn't be dialing. Is there a concrete algorithm for determining whether a phone number is normal. i.e. calling this phone number would result in a normal long distance rate. It seems like the pattern 1NXXNXX for the U.S. is fairly commonly used, but it wouldn't catch erroneous phone numbers such as 1411XXX (and the other X11 numbers) Is it just a matter of checking for these invalid area codes? If so, is there a list anywhere that I could check against? I tried googling this topic, but it's hard to find anything with such common keywords. If anyone can direct me to a good resource, I'd appreciate it as well. On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Thanks, Peter Hsu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Validating a phone number
On Sun, 2005-07-17 at 00:01 -0700, Peter Hsu wrote: I'm concerned about people dialing out of our asterisk server to numbers they shouldn't be dialing. Is there a concrete algorithm for determining whether a phone number is normal. i.e. calling this phone number would result in a normal long distance rate. It seems like the pattern 1NXXNXX for the U.S. is fairly commonly used, but it wouldn't catch erroneous phone numbers such as 1411XXX (and the other X11 numbers) normally those dont work with a 1 before them, but I cant say that is a guarantee with all providers. 19xx is normally premium service and has a sometimes steep charge. 1700xxx is another number that normally places a higher than normal charge to callers for calling. In america there are some numbers that appear normal but are premium service numbers, there are some in NJ that charge $5 to call in the 201 area code but they can exist in other states as well. You may want to filter numbers that would fit the 1NXX... format but arent in the US or Canada either. There was a company that had a number )I forget where somewhere in the caribean) that was part of the NANPA (ie 1NXX) but charged $2511/minute to callers. Because they are not in the US the FTC rules about declaring that it is a premium service number and the charges when first called do not apply. There are only a couple area codes 809 seems to come to mind but I cant guarantee that. In short you might investigate a phone company service blocker for premium service numbers and try your best to block what you can but it would be impossible for someone without SS7 network access to see what the rate of the call is since these numbers can hide virtually anywhere. I tried googling this topic, but it's hard to find anything with such common keywords. If anyone can direct me to a good resource, I'd appreciate it as well. NANPA manages all the numbers in the north american numbering plan, if memory serves their page is nanpa.org and they used to have rate center information available on their page for free that you can download (and you would need to parse it and continually get updates as new exchanges are allocated). On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Ahh glad you brought that up, see above. I think there are a couple of them, but I dont know off hand what they are.. try googling 'toll fraud 809' and see if that works. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
Carl Andersson wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that,using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan My problem turned out to be a termination problem. When using zaphfc together with other zap cards, it seems to be of importance in which order the drivers are loaded as well - At least in my case it would only work right if the X100P driver was loaded before the zaphfc driver. I have got verything working now, so if you have any questions you are more than welcome. You didn't write if you intended to use the ISDN cards to connect to ISDN lines, or if you wanted to create a setup like mine, with the card/cards in NT mode, acting as an ISDN switch of it's own. Hi Carl, Thanks for this confirmation. My setup is 2 HFC cards to 2 ISDN lines, so nope, not like yours. Cheers, Zoltan. -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pingtel hardphone config' requested
Title: Pingtel hardphone config' requested If you can in any way improve this page: http://voip-info.org/tiki-index.php?page=Pingtel+Hardphone please do. Thanks very much. Jason Sjobeck ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Vovida Loadbalancer
Dear All, I was trying to load balance between two asterisk servers using vovida.org loadbalancer, but when I was running it i faced the following problems: -When phones try to register the lpproxy gives the following message for reach phone trying to connect: Sticky header data is: Call-ID: [EMAIL PROTECTED] No proxies are up - can not send message to anyone (I start the process by ./lbproxy -proxy asterisk1 -proxy asterisk2 ) and the phne is unable to register - doing a sip debug on both asterisk boxes I can see this message on both of them: find_call: Call missing call ID from 'loadbalancerserver' Did anyone face this before or worked with both asterisk and vovida loadbalancer ? Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] beginners question about extension context
Rudolf Ladyzhenskii wrote: Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a given context Here are my contexts definitions: [from-sip] exten =101,1,Dial(SIP/phone1) exten =102,1,Dial(SIP/phone2) exten =103,1,Dial(SIP/phone3) [iax-user] exten=201,1,Dial(IAX2/phone4) exten=202,1,Dial(IAX2/phone5) If I try to call from IAX2 phone to say ext 102, I get request '[EMAIL PROTECTED]' does not exist I have tried to include iax-user in from-sip and I can make calls from SIP phones to IAX2 ones, but not the other way around. Now for an interesting bit. If I include from-sip in tthe iax-user, all is working fine -- I can make calls in any directions. If I try to do cross-include where one context is included into another and vise versa, IAX2 phone does not even register. Is there a better than include way to route calls between contexts? Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, From one newbie to another: Hope I don't mis-lead you. My understanding is that in your dialplan you have options to route calls based (amongst other things) on where they originate and where the want to terminate. To this end you should put all your internal phones on the same context (lets use from-internal for this example), regardless of whether thay are sip, iax or who cares. Note that sip.conf, iax.conf etc define the type of phone and will let Now, in your dialplan (extensions.conf) you set up a from-internal context and if anyone in that context dials 201, you route it to phone4, 101 you route to phone1 etc etc. This setup also allows you to trap external incoming calls and stop them from making an external outgoing call (that you will end up paying for). (If anyone thinks I've got this wrong, please comment.) HTH, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Read error om sound device
Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 Now I installed the bristuff+asterisk 1.0.9 and always have in my logs [chan_oss.so]: (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Read error on sound device: Resource temporarily unavailable Trying a dial 3 command in console (config file from scratch) give me BackGround(OSS/dsp,demo-congrats) [...] Timeout on OSS/dsp console Hangup /dev/dsp is a symlink to /dev/dsp0 owned by root/audio and crw-rw In a terminal, with mpg123, I played all the mohmp3 files without any problem. Also, connecting a UA and doing the same Call to extension 3 from there is working fine. Did someone else have this issue with 1.0.9? I try to follow whith strace -o, this error is even not notice there! Thanks for any hint. -- Daniel TOOTAI Networks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager API commands QueueStatus and Queues
Does anyone know what the descriptions are for the data that QueueStatus and Queues manager API commands return? Any information would be helpful. Thanks in advance. anything that I know about the events is in the javadocs of Asterisk-Java. Have a look at http://asterisk-java.sourceforge.net/apidocs/net/sf/asterisk/manager/event/package-summary.html If you get any further information about the attributes please let me know so I can include it. =Stefan signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Build from 16-7-2005 Crash! bug or what? ;-D
On Sun, 2005-07-17 at 04:08 +0200, xAD wrote: I have tried to update my CVS build from 29-6-2005 with a new one. but now when i start asterisk in verbose mode it crash after 1000+ lines of: Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-252000 ... i have rollback to the previous one (29-6-2005) and works perfectly. Confirmed here also I'm lucky if I can keep an X101P running without a red alarm for more than 30 mins. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c:5606 check_auth: stale nonce received from
I get from one user following entry in CLI chan_sip.c:5606 check_auth: stale nonce received from What does it mean, and do I need to do something? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xten does not want to dial
I sure have just something forgotten: I have setup a new phone user in Realtime sipbuddies, have enabled a dialing plan according to his context, ... I login with Xten and see it via sip show peer. I can call this number, voice, ... all seems ok. HOWEVER, I cannot call from Xten. What ever I dial, ... there is nothing to see on CLI Xten setup: Sip Proxy Disabled Default Enabled Proxy1 instead Setings in Proxy 1 Enabled: yes Username: 223344 Authorization User: 223344 Password: Domain/Realm: vpbx.elmit.com Sip Proxy: vpbx.elmit.com Out Bound Proxy Use Outbound Proxy: Default (rest default) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Log Analyser Build into IPS 0.123
IPSwitchBoard Version 0.123 - 17 July 2005 * Queue Analysis has been added to IPSwitchBoard. In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail than traditional call detail records provide. In order to support this, extensive and detailed tracing of every queued call is stored in the queue log, located (by default) in /var/log/asterisk/queue_log. A Queue Log Analyser has now been build into IPSwitchBoard. Start IPSwitchBoard and the log file is retrieved automatically from Asterisk and there's a lot of reports you can enjoy: 1. Answered/unanswered Calls by Date, Hour or Weekday 2. Service Level by Date, Hour or Weekday 3. Agent calls - who received all those calls 4. Unanswered call - detail report with caller ID If you need any other reports, please drop me an e-mail on [EMAIL PROTECTED] FREE Download: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implementing a ISDN home PBX
Arik Funke wrote: Hi, I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? No. Use the Fritz!-card for the external connection, but an HFC-card for the internal bus. (see http://www.voip-info.org and search for HFC) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?
[snip] as well as the software, while the Polycoms can be centrally managed via TFTP/FTP/HTTP/HTTPS, etc. You mean: getting close to almost barely good enough to be as managable as a local software? No, I mean, if I have 50 extensions, I can create one config file, arrange it however I need on the server, and manage the 50 phones from my desk via ssh, etc. With softphones, I will have to get up and walk to each desk to change settings if I need to. Unless your OS has a really really high TCO to manage, those hardware phones are much more of a pain at that point. Again, I think this is true if you have 1-5 phones, but if you have 50+, I think not! This isn't even considering that you might have phones in remote locations, such as one of our branch ofices that is 3+ hours away. Central phone config means that I can make a change at 8:00PM and all of my users will have received it when the offices open in the AM, but softphones means I would have to remind everyone to leave their PCs on so I could remotely change the software config via VNC, and I don't evenknow if I would have to worry about user profiles having different settings, which would introduce another level of complexity. Of course, I could then set up centrally managed PCs, a la LTSP, but that's more of an undertaking than most want! [snip] I agree, your boss will judge the system based on is experience with it. So don't skim on the quality if you want to keep him happy. This is why I think that it is worth the extra $50 or so for the cheaper hardphones. OTOH, there is the false logic that just because you didn't pay enough for it, means its quality is low. For instance, on typical mainframe installation, people spend much more on basically the same thing. This is because they've already payed the 1,000,000$ for the system, and are used to pay a bit more for accesories. Agreed, the you get what you pay for statement isn't always true, but I think it is, at least for most business situations, especially those with lower-tech workers (ie: not Power-users who will learn the special key shorcuts, etc.) [snip] Summary: I'm not sure soft phones are there yet, but I suspect they will be good enough for more and more people. I have to agree with you here, but I also think we'll have to agree to disagree on other points! Basically, it all depends on your situation, but for me, and I think that for most business users, small, medium, or large, the reasonable minor additional cost of a hardphone will be worth it. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Log Analyser Build into IPS
IPSwitchBoard Version 0.123 - 17 July 2005 * Queue Analysis has been added to IPSwitchBoard. In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail than traditional call detail records provide. In order to support this, extensive and detailed tracing of every queued call is stored in the queue log, located (by default) in /var/log/asterisk/queue_log. A Queue Log Analyser has now been build into IPSwitchBoard. Start IPSwitchBoard and the log file is retrieved automatically from Asterisk and there's a lot of reports you can enjoy: 1. Answered/unanswered Calls by Date, Hour or Weekday 2. Service Level by Date, Hour or Weekday 3. Agent calls - who received all those calls 4. Unanswered call - detail report with caller ID If you need any other reports, please drop me an e-mail on [EMAIL PROTECTED] FREE Download: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implementing a ISDN home PBX
On Sun, 17 Jul 2005, Arik Funke wrote: Hi, I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? What do you mean with 'internal bus'? If you mean an internal S0 of a ISDN-PBX, then yes, that will work. But if you mean to connect the phones directly to the FritzCard, then no. You need the card to operate in NT mode, which FritzCard cannot. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS SRV
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
As I understand it should mean that any sip connection to
anyname@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten = ronald,1,Dial(${PHONE_615},60,tr)
exten = ronald,2,Voicemail,[EMAIL PROTECTED]
exten = ronald,103,Voicemail,[EMAIL PROTECTED]
sip.conf
[general]
context=default
srvlookup=yes
reloaded
The person you are calling is not available
What do I miss?
bye
Ronald Wiplinger
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Re: [Asterisk-Users] DNS SRV
Ronald Wiplinger wrote:
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
Answering my own post:
a. delete the *.jnl of the dns server
b. use tabs instead of spaces around IN and SRV (didn't we make these
errors, when we were setting up DNS )
bye
Ronald Wiplinger
As I understand it should mean that any sip connection to
anyname@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten = ronald,1,Dial(${PHONE_615},60,tr)
exten = ronald,2,Voicemail,[EMAIL PROTECTED]
exten = ronald,103,Voicemail,[EMAIL PROTECTED]
sip.conf
[general]
context=default
srvlookup=yes
reloaded
The person you are calling is not available
What do I miss?
bye
Ronald Wiplinger
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Re: [Asterisk-Users] howto on ISDN HFC cards with AAH v1.1
Tzafrir Cohen wrote: On Sat, Jul 16, 2005 at 07:28:06PM +0200, Zoltan Szecsei wrote: Tzafrir Cohen wrote: On Sat, Jul 16, 2005 at 05:40:28PM +0200, Zoltan Szecsei wrote: Tzafrir Cohen wrote: Don't know about [EMAIL PROTECTED], but upcoming version of Rapid has support for zaphfc from bristuff. I also extended our zaptel detection script to support zaphfc . However it's still kind of untested. Want to test? Hi Tzafrir - you well? Sadly dont have time (hope it is not a short-sighted answer from me) - I have to get this box up and running by Monday afternoon. Here's the fun part - at my place I have adsl but no ISDN, at the (new) office where this system is going, they have ISDN but no (yet) access to the internet. So, I can read up install here, but I can only test (but not read up query) at the new office. This is also my first production asterisk effort. How do I get into these situations?? :-) What can I say? Once you get bristuff built, try my up-to-date genzaptelconf from http://tzafrir.org.il/genzaptelconf . The folks from AAH modified it a bit and I never bot any useful patches from them, soi I have no idea how this works with latest AAH. Thanks - will do. I think everythings just been stuffed up. The bristuff install shell wgets asterisk 1.0.8, zaptel libpri. then auto untars it and runs a compile script. bristuff RC8h is for 1.0.8, RC8j is for 1.0.9 . Apart from that, there's proctically no difference between the two. zaptel libpri got errors, but asterisk seems ok (i just ran it my phones are authenticating). So maybe you use old zaptel libpri ? Nope - remember: you make in the order of zaptel, libpri then asterisk, which siggetst at zaptel time, asterisk is not relevant. On looking at it, my actual problem was that there was no /usr/src/linux link to the headers (only a /usr/src/linux-2.4) - there is also a lovely rant about redhat in the README file - so all I had to do was: ln -s /usr/src/linux-2.4 /usr/src/linux ./compile.sh so now to work out what the f... about those damn isdn cards. :-( Do you have a spare partition on that system? If so, give rapid a shot... I wish!! AAH takes over the entire HD format it accordingly. I would love to have split this HD into 2 partition sets, so that I could install AAH on the one set and SuSE9.3 with latest asterisk on the other partition set. (I dont have time to play with parted and sort this out as this system needs to be up and running by tomorrow) Cheers for now, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read error om sound device
On Sun, Jul 17, 2005 at 10:47:29AM +0200, Administrator TOOTAI wrote: Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235 ]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 you use alsa, right? Now I installed the bristuff+asterisk 1.0.9 and always have in my logs [chan_oss.so]: (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Read error on sound device: Resource temporarily unavailable Why use OSS, then? Trying a dial 3 command in console (config file from scratch) give me BackGround(OSS/dsp,demo-congrats) [...] Timeout on OSS/dsp console Hangup /dev/dsp is a symlink to /dev/dsp0 owned by root/audio and crw-rw asterisk is in the group audio, right? groups asterisk If it is not, this is a bug. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?
On Sun, Jul 17, 2005 at 05:19:05AM -0400, Tom Rymes wrote: [snip] as well as the software, while the Polycoms can be centrally managed via TFTP/FTP/HTTP/HTTPS, etc. You mean: getting close to almost barely good enough to be as managable as a local software? No, I mean, if I have 50 extensions, I can create one config file, arrange it however I need on the server, and manage the 50 phones from my desk via ssh, etc. With softphones, I will have to get up and walk to each desk to change settings if I need to. This is software. Use manageble software. If software means separate setup on each desktop, then don't use it. If you spend that much time on setting up phones, imagine how long it takes you to update other software packages. This is, then, a symptom of a general problem. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple ISDN BRI Units with AsteriskusingBristuff zaphfc in NT mode?
The terminating resistors, or lack of, should not effect correct operation if bri circuits unless multiple devices are sharing the same bus. From experience even then are not essential unless it is a long line. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Snelders Sent: 16 July 2005 21:38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Multiple ISDN BRI Units with AsteriskusingBristuff zaphfc in NT mode? Try terminating using 50 ohm resistors as suggested by this guide: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html in chapter 2.2 (Connect ISDN telephones to your ISDN card.) Best regards, Jan Snelders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? Carl Andersson wrote: Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that, using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read error om sound device
Tzafrir Cohen a écrit : On Sun, Jul 17, 2005 at 10:47:29AM +0200, Administrator TOOTAI wrote: Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 you use alsa, right? Yes Now I installed the bristuff+asterisk 1.0.9 and always have in my logs [chan_oss.so]: (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Read error on sound device: Resource temporarily unavailable Why use OSS, then? OSS emulation of alsa. No need to build chan_alsa which needs alsa sources Trying a dial 3 command in console (config file from scratch) give me BackGround(OSS/dsp,demo-congrats) [...] Timeout on OSS/dsp console Hangup /dev/dsp is a symlink to /dev/dsp0 owned by root/audio and crw-rw asterisk is in the group audio, right? groups asterisk If it is not, this is a bug. Yes, asterisk is in group audio. groups asterisk asterisk : users audio Running asterisk this way (under asterisk user) [chan_oss.so] = (OSS Console Channel Driver) Unable to open /dev/dsp: Permission denied == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'no-load=chan_oss.so' in /etc/asterisk/modules.conf Anyone else is using 1.0.9 with dial command in CLI? -- Daniel TOOTAI Networks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read error om sound device
On Sun, Jul 17, 2005 at 02:44:27PM +0200, Administrator TOOTAI wrote: Tzafrir Cohen a écrit : On Sun, Jul 17, 2005 at 10:47:29AM +0200, Administrator TOOTAI wrote: Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: Have you installed the version from Sarge, or built one on your own? cat /proc/assound/cards 0 [V8235]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 you use alsa, right? Yes Now I installed the bristuff+asterisk 1.0.9 and always have in my logs [chan_oss.so]: (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Read error on sound device: Resource temporarily unavailable Why use OSS, then? OSS emulation of alsa. No need to build chan_alsa which needs alsa sources Trying a dial 3 command in console (config file from scratch) give me BackGround(OSS/dsp,demo-congrats) [...] Timeout on OSS/dsp console Hangup /dev/dsp is a symlink to /dev/dsp0 owned by root/audio and crw-rw asterisk is in the group audio, right? groups asterisk If it is not, this is a bug. Yes, asterisk is in group audio. groups asterisk asterisk : users audio That's strange. isn't asterisk in the group asterisk? is its primary group 'users'? Running asterisk this way (under asterisk user) [chan_oss.so] = (OSS Console Channel Driver) Unable to open /dev/dsp: Permission denied == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'no-load=chan_oss.so' in /etc/asterisk/modules.conf Anyone else is using 1.0.9 with dial command in CLI? No. But http://www.pbxfreeware.com/app_changrab.c has one (originate). Should probably build just fine with asterisk-dev installed, but let me know if you need a skeleton deb. As for me, I never bothered, as I use a script to generate a call file and thus have basically the same functionality. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Validating a phone number
I think the address is http://nanpa.com/ nanpa.org is photography site :) On 7/17/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: NANPA manages all the numbers in the north american numbering plan, ifmemory serves their page is nanpa.org and they used to have rate centerinformation available on their page for free that you can download (andyou would need to parse it and continually get updates as new exchanges are allocated). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Have some latency problems.
Hi guys i have some problems with asterisk latency. I am trying to play online radio streaming on musiconhold and i am receiving bad quality of the sound and the latency, then i am calling from ip phone there is no problem but when i am calling from regular phone to my Asterisk server i receving this stuff: RFC3389: 1 bytes, level 4... RFC3389: 1 bytes, level 4... RFC3389: 1 bytes, level 4... RFC3389: 1 bytes, level 4... RFC3389: 1 bytes, level 4... maybe the problem in the codec or if there any way to reduce the size of the packets. Any help will be appreciated. Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
On Tuesday 12 July 2005 03:15 am, Sergio Chersovani wrote: Marc Fishman ha scritto: I appreciate the response but that's what isn't working. I have tried v5.3 and v3.0 with the same result. I suspect the firmware version (P003AM30) is I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Thanks for the response and apologies for the delay in answering. I got it squared away but it took seven firmware upgrades (in the correct order) to get there. For anyone interested, the following firmware upgrade path worked for me: Cisco 7960 starting from factory installed P003AM30 P0S30200 P0S30203 P0S30300 P0S3-05-3-00 P0S3-06-3-00 P003-07-1-00 P0S3-07-2-00 -- Marc H. Fishman OuttaSite Resources If you woke up breathing, congratulations! You have another chance! PGP KeyID: 6C8E212E75CDBD79 PGP Key Fingerprint: E620 1F11 D3AC 6FEC 4CC5 8CA6 6C8E 212E 75CD BD79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Interface with mobile phone
Thanks; I willcheck out the internal devices and the cellsockets like devices hopefully i can get the ones that are compatible with Asterisk. Mr Wehr mentioned gsm gateway, what is it and how does't work? I am posting the question and I will also do my searching homework. Regards; "trixter http://www.0xdecafbad.com" [EMAIL PROTECTED] wrote: On Sat, 2005-07-16 at 11:55 -0700, chawki hammoud wrote: Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phones can be differntaited by their prefix. Thanks There are a few ways to do this.. chan_bluetooth is one if you have amobile that speaks bluetooth. www.cellsocket.com if you have acompatible phone. And some others have previously mentioned (and Iforgot, the arc hives have it or maybe the posters would be nice andrepost) devices that work like cellsocket. Basically cellsocket andsimilar devices are a charging base station for mobiles that provide aFXS port on the mobile side. Connect it to an FXO port and you can useit as a normal phone. Additionally there are internal devices that work directly as a mobilebut is fixed in the computer (pccard or usb typically) if you dont havethe need to remove the mobile and take it with you when you arent athome/office.-- Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Interface with mobile phone
Hi, On 06:48, Sun 17 Jul 05, chawki hammoud wrote: Thanks; I willcheck out the internal devices and the cellsockets like devices hopefully i can get the ones that are compatible with Asterisk. Mr Wehr mentioned gsm gateway, what is it and how does't work? I am posting the question and I will also do my searching homework. Regards; There's this device called VoiceBlue GSM gateway. It talks gsm on one side and SIP on the other side. Have a look at: http://www.voip-info.org/tiki-print.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] wrote: On Sat, 2005-07-16 at 11:55 -0700, chawki hammoud wrote: Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phones can be differntaited by their prefix. Thanks There are a few ways to do this.. chan_bluetooth is one if you have a mobile that speaks bluetooth. www.cellsocket.com if you have a compatible phone. And some others have previously mentioned (and I forgot, the archives have it or maybe the posters would be nice and repost) devices that work like cellsocket. Basically cellsocket and similar devices are a charging base station for mobiles that provide a FXS port on the mobile side. Connect it to an FXO port and you can use it as a normal phone. Additionally there are internal devices that work directly as a mobile but is fixed in the computer (pccard or usb typically) if you dont have the need to remove the mobile and take it with you when you arent at home/office. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Validating a phone number
On Sun, Jul 17, 2005 at 12:15:06AM -0700, trixter http://www.0xdecafbad.com wrote: In short you might investigate a phone company service blocker for premium service numbers and try your best to block what you can but it would be impossible for someone without SS7 network access to see what the rate of the call is since these numbers can hide virtually anywhere. The cost of a call is not available from any SS7 service that I know of. NANPA manages all the numbers in the north american numbering plan, if memory serves their page is nanpa.org and they used to have rate center information available on their page for free that you can download (and you would need to parse it and continually get updates as new exchanges are allocated). http://www.nanpa.com/area_codes/index.html You'll have to make a white list. On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Ahh glad you brought that up, see above. I think there are a couple of them, but I dont know off hand what they are.. try googling 'toll fraud 809' and see if that works. 809 is a valid area code. Read this: http://www.lincmad.com/telesleaze.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse connect - unable to dial out, asterisk says 9696
Thanks, I had added more funds a couple weeks ago, however I just found this on the voicepulse connect KB:- Why do I hear error message '9696' when dialing with Asterisk? Question Why do I hear error message '9696' when dialing with Asterisk? Answer When making outgoing calls using the VoicePulse Connect! service, if you hear a message stating 9696, your account has been suspended for negative balance or misuse. Please contact technical support through the Account Center to resolve the issue. So it looks like even though I added $20 credit, I'm still suspended. Anyway I contacted them, lets see what happens. Mike On 7/17/05, S. William Schulz [EMAIL PROTECTED] wrote: Mike Dent wrote: -- Called NBhX:[EMAIL PROTECTED]/12124565900 -- Call accepted by 66.234.228.160 (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/66.234.228.160:4569/1' -- Executing HasNewVoicemail(SIP/2008-cf55, 2002) in new stack rt*CLI and Asterisk speaks back to me 96 96 I seem to recall a similar error mentioned and the problem turned out to be a need to add more funds to the account. I can't say that this is the case, but it might be worth checking. S ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Vovida Loadbalancer
On Sunday 17 July 2005 04:28, Mohamed A. Gombolaty wrote: Dear All, I was trying to load balance between two asterisk servers using vovida.org loadbalancer, but when I was running it i faced the following problems: What happend here is that you hijacked someone elses thread. Your email program probably does not work properly with threads, and you thought that if you simple change the subject all is fine. The tread information is however a subject in the header, and it does not matter how much you change the subject, it's still the same thread. What I suggest you do is to right click on the email address and select new, this way you won't interrupt others, and your chances of getting replies to your email increases. -- List Manager Network Voice Comunications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implementing a ISDN home PBX
Armin Schindler wrote: On Sun, 17 Jul 2005, Arik Funke wrote: I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? But if you mean to connect the phones directly to the FritzCard, then no. You need the card to operate in NT mode, which FritzCard cannot. I meant indeed to connect phones directly to the Fritz card. I found a list of cards that support nt mode with mISDN: http://isdn.jolly.de/cards.html Would any of these work or do they only work with mISDN but not with Asterisk for my application? Sorry if this question does not make sense. I am not yet sure how Asterisk handles ISDN cards. Especially with the two capi channels. What is the difference between the two capi channel drivers for asterisk? Which one do I use for what? Or are they interchangeable? Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] beginners question about extension context
Rudolf Ladyzhenskii wrote: Is there a better than include way to route calls between contexts? [internal-extensions] include = from-sip include = iax-users [from-sip] include = internal-extensions [iax-users] include = internal-extensions Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?
On Sunday 17 July 2005 05:19, Tom Rymes wrote: [snip] away. Central phone config means that I can make a change at 8:00PM and all of my users will have received it when the offices open in the AM, but softphones means I would have to remind everyone to leave their PCs on so I could remotely change the software config via VNC, and I don't You can (often) get around that by having each client automatically check for updates when the user logs in. I had a batch file for each user that looked for a file and then executed it if found. I got a lot of changes done this way. Of course your milage may vary, but it's amazing what you can do with a bit of ingenuity. -- List Manager Network Voice Comunications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing via sipgate - remote answer does not stop asterisk internal ring until cycle finished?
Hi, Sipgate works, I can dial out fine. When I make a SIP call with sipgate (to pstn) I am presented with the ring tone (US style) from *. If the remote end answers at the start of one of these ring cycles from asterisk, it does not interrupt the ring and the remote end just hears silence until that ring cylce finishes, at which time the call is connected. It sometimes causes the remote end to hangup as they think nobody is there. Am I missing something? thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323.conf ... how to regitster users ... tell me PLZZZZZZ
Assalam Alaikum i m using oh323.conf n im calling netmeeting/SJPhoneusing Dial(oh323/IP address of netmeeting or anysoftphone) how can i call any extension first of all tell me how to register a uid n password there let's say i've a user type=friend username=adeel secret=adeel context=incoming mailbox=31 plz plz plzz send me just a sample oh323.conf n related portion of extesions.conf containg above info Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipjet test account - unable to make calls.
Hi, I just setup a VoipJet test account (one with 25c credit) to test, they seem to offer good rates to 02 Uk mobiles :) Anyway, everything went ok, iax.conf amended and extensions.conf too, however when I try to make a call I see:- rt*CLI -- Executing SetCallerID(SIP/2008-d747, 4153574000) in new stack -- Executing Dial(SIP/2008-d747, IAX2/[EMAIL PROTECTED]/01144524592060) in new stack -- Called [EMAIL PROTECTED]/01144524592060 Jul 17 17:04:37 WARNING[31624]: chan_iax2.c:5553 socket_read: Call rejected by 66.246.220.19: No authority found -- Hungup 'IAX2/voipjet/1' == No one is available to answer at this time Do these accounts take a while to become 'active' ? I think it's about 1 hour now since I set it up. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implementing a ISDN home PBX
On Sun, 17 Jul 2005, Arik Funke wrote: Armin Schindler wrote: On Sun, 17 Jul 2005, Arik Funke wrote: I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? But if you mean to connect the phones directly to the FritzCard, then no. You need the card to operate in NT mode, which FritzCard cannot. I meant indeed to connect phones directly to the Fritz card. I found a list of cards that support nt mode with mISDN: http://isdn.jolly.de/cards.html Would any of these work or do they only work with mISDN but not with Asterisk for my application? Sorry if this question does not make sense. I am not yet sure how Asterisk handles ISDN cards. Especially with the two capi channels. I cannot tell you anything about the mISDN status. But as far as I know, when you want to use easy and cheap NT-mode, the zaphfc cards will do very good. What is the difference between the two capi channel drivers for asterisk? Which one do I use for what? Or are they interchangeable? I believe you mean chan_capi from junganns and my chan_capi-cm version on sourceforge? Well, they are not really different. chan_capi-cm is just ongoing development, newer versions, fixes, new features Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c:5606 check_auth: stale nonce received from
Ive been getting these too. Mainly from my Linksys-PAP2NA users. -Matthew From: Ronald Wiplinger [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 17 Jul 2005 17:24:03 +0800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_sip.c:5606 check_auth: stale nonce received from I get from one user following entry in CLI chan_sip.c:5606 check_auth: stale nonce received from What does it mean, and do I need to do something? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
All, The thorny old issue of upgrading 7940/7960s again... here's the abridged version of the info I've produced before: a) Cisco's documentation is buggy - and I've told them! b) You don't need an OS79XX.TXT if you use my upgrade route c) I recommend that you upgrade a 7940/7960 which is in an unknown state to SCCP/Skinny version P00306000100 ie. 6.0(1) first then then over to the version of SIP you want, eg. 7.5(0) which is now current. This approach gets all the right bits in place for signed binary support and minimises the number of flash loads you need (and saves time :o) d) attached is a little program (phonecfg.c) which is 70% documentation 30% code read it, and use it, for an easy life... do the following: - put relevant files and the program in your TFTP boot directory - make an XML config file for SCCP operation: ./phonecfg -c macaddress - boot the phone and watch it upgrade - make an XML config file for SIP operation ./phonecfg -s macaddress - boot the phone and watch it upgrade I worked out what is _really_ going on using TCP dump and fixed it from the ground up, rather than relying on buggy documentation. This approach works for me(tm) and will probably work for you too. Regards Mike - Original Message - From: Marc Fishman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 17, 2005 2:30 PM Subject: Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems On Tuesday 12 July 2005 03:15 am, Sergio Chersovani wrote: Marc Fishman ha scritto: I appreciate the response but that's what isn't working. I have tried v5.3 and v3.0 with the same result. I suspect the firmware version (P003AM30) is I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Thanks for the response and apologies for the delay in answering. I got it squared away but it took seven firmware upgrades (in the correct order) to get there. For anyone interested, the following firmware upgrade path worked for me: Cisco 7960 starting from factory installed P003AM30 P0S30200 P0S30203 P0S30300 P0S3-05-3-00 P0S3-06-3-00 P003-07-1-00 P0S3-07-2-00 -- Marc H. Fishman OuttaSite Resources If you woke up breathing, congratulations! You have another chance! PGP KeyID: 6C8E212E75CDBD79 PGP Key Fingerprint: E620 1F11 D3AC 6FEC 4CC5 8CA6 6C8E 212E 75CD BD79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users phonecfg.c Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read error om sound device
Tzafrir Cohen a écrit : On Sun, Jul 17, 2005 at 02:44:27PM +0200, Administrator TOOTAI wrote: Tzafrir Cohen a écrit : On Sun, Jul 17, 2005 at 10:47:29AM +0200, Administrator TOOTAI wrote: Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: Have you installed the version from Sarge, or built one on your own? As I told in first mail, Sarge version (deb) was installed and had no problem, but it's a 1.0.7 version, I want to have the latest stable. [...] No. But http://www.pbxfreeware.com/app_changrab.c has one (originate). Should probably build just fine with asterisk-dev installed, but let me know if you need a skeleton deb. As for me, I never bothered, as I use a script to generate a call file and thus have basically the same functionality. I saw you have build problems. I will try it but I really want to know why it's not working. If it's a bug or something else. For my knowledge ;-) -- Daniel TOOTAI Networks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo fails to find Sweex CA000022 - X100P clone
Hi, I'm trying to figure out why my Sweex CA22 is not detected by wcfxo on my SuSE Pro 9.3 running on a VIA EPIA SP motherboard. The card should work as its an Intel 82537EP based card. When Booting the IRQ assignments for the board indicate: Bus Dev Func Vendor Dev Class 020 0 8086 1080 0703 lspci shows: :00:14.0 Modem: Intel Corporation: Unknown device 1080 (rev 04) (prog-if 00 [Generic]) Subsystem: Intel Corporation: Unknown device 1000 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 15500ns max), cache line size 08 Interrupt: pin A routed to IRQ 11 Region 0: Memory at ee003000 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at ec00 [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- But when performing a modprobe I get: linux:/usr/src/zaptel # modprobe -v wcfxo insmod /lib/modules/2.6.11.4-21.7-default/kernel/lib/crc-ccitt.ko insmod /lib/modules/2.6.11.4-21.7-default/misc/zaptel.ko install /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg insmod /lib/modules/2.6.11.4-21.7-default/misc/wcfxo.ko ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo Based on the info shown by the BIOS, can wcfxo correctly detect this card or does it require changes to wcfxo_pci_tbl ? If so, what values need to be added? I want to get this card working so I can document the process and details on voip-info for others to use. Many thanks Richard Tomlinson _ http://www.sysgen.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read error om sound device
On Sun, Jul 17, 2005 at 07:19:15PM +0200, Administrator TOOTAI wrote: Tzafrir Cohen a écrit : On Sun, Jul 17, 2005 at 02:44:27PM +0200, Administrator TOOTAI wrote: As I told in first mail, Sarge version (deb) was installed and had no problem, but it's a 1.0.7 version, I want to have the latest stable. That's what backports are for. Consider what you'll have to upgrade to apply the next critical security fix. [...] No. But http://www.pbxfreeware.com/app_changrab.c has one (originate). Should probably build just fine with asterisk-dev installed, but let me know if you need a skeleton deb. As for me, I never bothered, as I use a script to generate a call file and thus have basically the same functionality. I saw you have build problems. I will try it but I really want to know why it's not working. If it's a bug or something else. For my knowledge ;-) Not sure exactly. It seems that the code is for HEAD. And I didn't really bother much trying. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe wcfxo fails.
I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delete voicemail
On Thu, Apr 28, 2005 at 09:43:23AM -0700, Wiley Siler wrote: Command line on the box and navigate to the directory for your VM. An example of one of mine... /var/spool/asterisk/voicemail/default/1003/INBOX/ Issue the rm *.* command A petty note: 'rm *', actually . rm *.* won't delete files that happen not to have '.' in their name. Also note that wildcards behave quite differently on OS and in unix shells. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please, excuse me
Title: Mensaje I'm sorry for my holidays message, but I think it's too hard span me from list, don't you think? Could admin return to list, please? Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe wcfxo fails.
Hey Tim, I'll be glad to help you out if I am able to.. but I honestly don't recall which thread you are talking about. Maybe if you refresh my mind and/or explain your problem? Leandro Tim King wrote: I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours
Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten. This meant that when you dialed an extension which played back these files, the video wouldn't start unless you sent video first. Now, if you don't click send video in Xten until after you hear the beep for recording, it works. It will play you the video just by dialing the extension. I'm using the latest beta of eyebeam by the way. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC BRIstuff woes
Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] = (Zapata Telephony w/PRI) Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Not found (No such file or directory) :-( Jul 17 19:57:54 WARNING[2503]: Unable to specify channel 1: Device or resource busy :-( Jul 17 19:57:54 ERROR[2503]: Unable to open channel 1: Device or resource busy :-( here = 0, tmp-channel = 1, channel = 1 :-( Jul 17 19:57:54 ERROR[2503]: Unable to register channel '1-2' :-( Jul 17 19:57:54 WARNING[2503]: chan_zap.so: load_module failed, returning -1 :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Tor' :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Zap' :-( Jul 17 19:57:54 WARNING[2503]: Loading module chan_zap.so failed! Here's some background: Loaded a new AAH 1.1 Centos 3.4 on a PIII with 2 pci HFC ISDN - Asterisk loaded I could play with internal extensions. Added bristuff-0.2.0-RC8h which did a wget and updated to asterisk 1.0.8 - all seemed OK I could load asterisk. Did make load manually (by running the steps in zaphfc/Makefile) and ztcfg returned: [EMAIL PROTECTED] zaphfc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) [EMAIL PROTECTED] zaphfc]# The original /etc/zaptel.conf was empty so I added the one from zaphfc/ The original /etc/asterisk/zapata had a whole lot of stuff in it so I *patched in* the details from zaphfc/zapata.conf So, I'm looking for help on what's gone wrong. Couple of thoughts: Before I added the bristuff I had a zap/g0 channel - given I had only 2 IAX IP phones and an x-lite SIP softphone from my XP box, is this channel even used? There are 2 HFC cards in the box - yet zaptel.conf only has: span=1,1,3,ccs,ami bchan=1-2 dchan=3 Surely that only relates to one of the ISDN cards? Here is a listing of my (hybrid) zapata.conf that I patched with the bristuff zaphfc/zapata.conf. Can anyone tell me what I should have deleted - I am nervous to mess up all the facilities that came preconfigured with AAH1.1, but I'm sure some of the lines will clash with the ISDN needed stuff. What should/must I delete? TIA, Zoltan. [EMAIL PROTECTED] asterisk]# cat zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;start zoltan changes ; zoltan commented out: signalling=fxs_ks ; zoltan commented out: rxwink=300; Atlas seems to use long (250ms) winks signalling=bri_cpe_ptmp ; added by zoltan from bristuff p2mp TE mode ;signalling=bri_cpe ; added but commented by zoltan from bristuff if p2p TE mode ; end zoltan changes - but see lower down ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ; START added by zoltan from bristuff pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 09 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 ; END added by zoltan from bristuff ;Include AMP configs #include zapata_additional.conf ; this file is empty on zoltan pc ;Include genzaptelconf configs #include zapata-auto.conf ; file does not exist on zoltan pc [EMAIL PROTECTED] asterisk]# -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk -Users] modprobe wcfxo fails.
I don't remember, what was the problem? Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 9:09 AM Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 2 asterisks connected but trying to bridge
After adding notransfer=yes to both asterisks Im still getting this on mine Channel (ContextExtensionPri ) State Appl. Data SIP/intruder1-0462 (telefonos 1 ) Up Bridged Call IAX2/[EMAIL PROTECTED] IAX2/[EMAIL PROTECTED] (macro-stdexten s106 ) Up Dial SIP/intruder1|20|mwtWT Also, on the remote side I get this Jul 15 21:34:28 WARNING[850]: chan_sip.c:1047 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] Any ideas? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Vincent Luba |Sent: Viernes, 15 de Julio de 2005 03:19 a.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] RE: 2 asterisks connected but trying |to bridge | |Hey, |For the bridge issue, take a look at 'notransfer=yes' option |in your iax.conf. |It'll force * to stay in the path |http://www.mail-archive.com/asterisk-users@lists.digium.com/msg 42262.html | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 2 asterisks connected but trying to bridge
Also, both asterisks have notransfer?yes and I get this -- Attempting native bridge of IAX2/[EMAIL PROTECTED] and IAX2/voipjet-9 Why? Seems its not taking the notransfer into account. Now Im puzzled |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Vincent Luba |Sent: Viernes, 15 de Julio de 2005 03:19 a.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] RE: 2 asterisks connected but trying |to bridge | |Hey, |For the bridge issue, take a look at 'notransfer=yes' option |in your iax.conf. |It'll force * to stay in the path |http://www.mail-archive.com/asterisk-users@lists.digium.com/msg 42262.html | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC BRIstuff woes
Sorry to reply to my own post!!! - but I have solved the immediate problem. ztdummy was still being loaded which is why zaphfc would not load. I basically haccked etc/modules so that zaptel loads before zaphfc and ztcfg runs after loading zaphfc. I then hacked /etc/init.d/zaptel so that ztdummy was not loaded and zaphfc was. viola!! asterisk1*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodemoen 1demoen 2demoen asterisk1*CLI Does this mean that *both* the ISDN cards are seen ready for use ??? Please could someone check my hybrid zapata.conf at the end of this email ??? Cheers thanks, Zoltan. Zoltan Szecsei wrote: Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] = (Zapata Telephony w/PRI) Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Not found (No such file or directory) :-( Jul 17 19:57:54 WARNING[2503]: Unable to specify channel 1: Device or resource busy :-( Jul 17 19:57:54 ERROR[2503]: Unable to open channel 1: Device or resource busy :-( here = 0, tmp-channel = 1, channel = 1 :-( Jul 17 19:57:54 ERROR[2503]: Unable to register channel '1-2' :-( Jul 17 19:57:54 WARNING[2503]: chan_zap.so: load_module failed, returning -1 :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Tor' :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Zap' :-( Jul 17 19:57:54 WARNING[2503]: Loading module chan_zap.so failed! Here's some background: Loaded a new AAH 1.1 Centos 3.4 on a PIII with 2 pci HFC ISDN - Asterisk loaded I could play with internal extensions. Added bristuff-0.2.0-RC8h which did a wget and updated to asterisk 1.0.8 - all seemed OK I could load asterisk. Did make load manually (by running the steps in zaphfc/Makefile) and ztcfg returned: [EMAIL PROTECTED] zaphfc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) [EMAIL PROTECTED] zaphfc]# The original /etc/zaptel.conf was empty so I added the one from zaphfc/ The original /etc/asterisk/zapata had a whole lot of stuff in it so I *patched in* the details from zaphfc/zapata.conf So, I'm looking for help on what's gone wrong. Couple of thoughts: Before I added the bristuff I had a zap/g0 channel - given I had only 2 IAX IP phones and an x-lite SIP softphone from my XP box, is this channel even used? There are 2 HFC cards in the box - yet zaptel.conf only has: span=1,1,3,ccs,ami bchan=1-2 dchan=3 Surely that only relates to one of the ISDN cards? Here is a listing of my (hybrid) zapata.conf that I patched with the bristuff zaphfc/zapata.conf. Can anyone tell me what I should have deleted - I am nervous to mess up all the facilities that came preconfigured with AAH1.1, but I'm sure some of the lines will clash with the ISDN needed stuff. What should/must I delete? TIA, Zoltan. [EMAIL PROTECTED] asterisk]# cat zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;start zoltan changes ; zoltan commented out: signalling=fxs_ks ; zoltan commented out: rxwink=300; Atlas seems to use long (250ms) winks signalling=bri_cpe_ptmp ; added by zoltan from bristuff p2mp TE mode ;signalling=bri_cpe ; added but commented by zoltan from bristuff if p2p TE mode ; end zoltan changes - but see lower down ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ; START added by zoltan from bristuff pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 09 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 ; END
Re: [Asterisk-Users] HFC BRIstuff woes
On Sun, Jul 17, 2005 at 09:55:57PM +0200, Zoltan Szecsei wrote: Hi All, It's broken !! (drat) Naturally this is a bug in zaptel/bristuff/whatever, right? Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] = (Zapata Telephony w/PRI) Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata_additional.conf': Found Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Jul 17 19:57:54 VERBOSE[2503]: == Parsing '/etc/asterisk/zapata-auto.conf': Not found (No such file or directory) :-( Jul 17 19:57:54 WARNING[2503]: Unable to specify channel 1: Device or resource busy :-( Jul 17 19:57:54 ERROR[2503]: Unable to open channel 1: Device or resource busy :-( here = 0, tmp-channel = 1, channel = 1 :-( Jul 17 19:57:54 ERROR[2503]: Unable to register channel '1-2' :-( Jul 17 19:57:54 WARNING[2503]: chan_zap.so: load_module failed, returning -1 You have a problem with the definitions of channels 1,2 in zapata.conf . This stops asterisk from starting. :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Tor' :-( Jul 17 19:57:54 VERBOSE[2503]: == Unregistered channel type 'Zap' :-( Jul 17 19:57:54 WARNING[2503]: Loading module chan_zap.so failed! Here's some background: Loaded a new AAH 1.1 Centos 3.4 on a PIII with 2 pci HFC ISDN - Asterisk loaded I could play with internal extensions. Added bristuff-0.2.0-RC8h which did a wget and updated to asterisk 1.0.8 - all seemed OK I could load asterisk. Did make load manually (by running the steps in zaphfc/Makefile) and Do you use the patched versions of zaptel, libpri annd asterisk? ztcfg returned: [EMAIL PROTECTED] zaphfc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Can't configure the zaptel channel. Fix this before trying to load chan_zap What relevant cards do you have, exactly? cat /proc/zaptel/1 , please [EMAIL PROTECTED] zaphfc]# The original /etc/zaptel.conf was empty so I added the one from zaphfc/ The original /etc/asterisk/zapata had a whole lot of stuff in it so I *patched in* the details from zaphfc/zapata.conf So, I'm looking for help on what's gone wrong. Couple of thoughts: Before I added the bristuff I had a zap/g0 channel - given I had only 2 IAX IP phones and an x-lite SIP softphone from my XP box, is this channel even used? There are 2 HFC cards in the box - yet zaptel.conf only has: span=1,1,3,ccs,ami bchan=1-2 dchan=3 Surely that only relates to one of the ISDN cards? Here is a listing of my (hybrid) zapata.conf that I patched with the bristuff zaphfc/zapata.conf. Can anyone tell me what I should have deleted - I am nervous to mess up all the facilities that came preconfigured with AAH1.1, but I'm sure some of the lines will clash with the ISDN needed stuff. What should/must I delete? zapata.conf is not (yet) relevant here. No matter what channels you define in zapata.conf, they first must be configured by ztcfg. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Darren Wiebe wrote Did you do a make clean? I just, as in 1 hour ago, successfully installed 1.0.9 using the port on FreeBSD. Yeah, even deleted all the files in the asterisk ports , and refreshed it ports collection. Always fails to compile at this point. Am I missing a package dependency somewhere? Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours
Matt Riddell wrote: Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten. This meant that when you dialed an extension which played back these files, the video wouldn't start unless you sent video first. Now, if you don't click send video in Xten until after you hear the beep for recording, it works. It will play you the video just by dialing the extension. I'm using the latest beta of eyebeam by the way. Matt, Where did you get teh beta from? I purchased the release version but I would like to have the same version as you to get it working wth Asterisk. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Florz patch for zaphfc
Hi Nils, I would be interested in your hardware setup for the 4 cards, as I'm trying to do exactly the same. Two cards work fine in an Intel P2 System, while not even one of the Billion cards work in my AMD Athlon 2400+ (Via) Chipset System (pci performance to low - you might have some cpu throttling enabled ). Maybe it's just the board/bios etc. but, anyway I would appreciate your hardware recommendation. yours, Alex Nils Segerdahl wrote: On Sun, 23 Jan 2005, Stuart Hirst wrote: Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Im running bristuff-0.2.0-rc2b with Florians patch. 4 Billion hfc cards in ptp mode. Works like a charm. Even spandsp for receiving faxes works. Pelase describe your problem in more detail. /Nils Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- Jan 24 Eskimo Pie patented by Christian Nelson, 1922 Jan 24 Gold discovered in California at Sutter's Mill, 1848 Jan 24 DG Nova introduced, 1969 --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC BRIstuff woes
On Sun, Jul 17, 2005 at 10:49:24PM +0200, Zoltan Szecsei wrote: Sorry to reply to my own post!!! - but I have solved the immediate problem. ztdummy was still being loaded which is why zaphfc would not load. I basically haccked etc/modules so that zaptel loads before zaphfc and ztcfg runs after loading zaphfc. I then hacked /etc/init.d/zaptel so that ztdummy was not loaded and zaphfc was. viola!! asterisk1*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodemoen 1demoen 2demoen asterisk1*CLI Does this mean that *both* the ISDN cards are seen ready for use ??? This is just one ISDN card (its two channels). Please could someone check my hybrid zapata.conf at the end of this email ??? If you want a more interactive medium, try the IRC. Though you should have picked some time where more europeans are not asleep :-( -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
HI Hartmut, I do have the same problem as you decribed earlier. The Billion HFC Cards (two of them) work flawlessly in my old Pentium II, but in my more powerful Athlon XP 2400+ (Via KT400 Chipset), I allways get the pci performance too low message and syslog kills the system. I'll try it now with commenting out the message. Do you know anything more in the meantime. Did Mr. Junghanns have any statement about it? I wonder if chan_mISDN is a better choice by now? Thanks for any advice you might have! yours, Alexander Hartmut Wahl wrote: Hello, I have investigated the issue a bit further, I was not able to find the root cause, maybe it is the KT133 Chipset of my ASUS A7V. However I found a bad hack to make it work under some circumstances. I commented out the line: printk(KERN_CRIT zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.\n); in zaphfc/zaphfc.c. Since when this situation happened once, the syslog started and this caused the situation again - endless loop with high sysload. Now the sync lost happens probably every now and then but it does not go into an endless loop. I also recognized that I must not run setiathome since then the audio quality of connections via the hfc-card will suffer (crackling). Amazingly cpuburn (takes every cpu-Time it gets as well) does not have this effect. I have no idea what strange things setiathome does to cause this but it reminds me of a sound card problem on this board. My SB-Live had crackling when I ran setiathome but I think with a newer driver (and ALSA I think) this problem was gone. Things that did not help: - Trying to change the latency (is fixed to 16 and cannot be changed) - Trying to change the latency of my other pci-devices (much higher and much lower). - Playing around with BIOS Options like delayed transaction, etc. - Changing PCI-slots - Making sure that the ISDN-card did not share the IRQ - Having only the ISDN-card and the Video-card in the System Hope that helps others who encounter this problem as well Hartmut On Sat, Sep 11, 2004 at 12:55:21PM +0200, Hartmut Wahl wrote: Hi! On Sat, Sep 11, 2004 at 10:38:48AM +0200, Maurizio Marini wrote: my machine did hangup as growing logs fullfilled partition hmm I see, mine is 8G, but it has gronwn from 0.5G to 2.0G it does apply to asterisk, not to zaphfc :( it was a misleading suggestion, so i solved it installing in an other more powerful machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 999.556 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1957.88 with this hw i've no issues at all; even strange messages i complained about in my previous posts like: ok we are getting closer, although speed shouldn't be a problem: vendor_id : AuthenticAMD model name : AMD Athlon(tm) Processor stepping: 4 cpu MHz : 1059.618 cache size : 256 KB bogomips: 2097.15 it is a 1,4GHz underclocked, since my ASUS A7V with KT133 does only support 100MHz FSB, but it requires only slow and quite fans. I think I'll try different kernels maybe something in the Debian k7-kernel-image interferes. Greetings Hartmut ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours
Chris Mason (Lists) wrote: Matt, Where did you get teh beta from? I purchased the release version but I would like to have the same version as you to get it working wth Asterisk. From their forums: http://support.xten.net/viewtopic.php?t=3805 Don't forget that Asterisk will only record 263, not 263+ or 261,but can passthrough those. H263+ Looks really nice, it would be great if it could be used for playback. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe wcfxo fails.
Tim, Yes, it was resolved but cannot remember how... I'm using RedHat FC3, Asterisk 1.0.9 and it all just works assuming that you: - RTFM with respect to udev setup - use "make linux26" to build the kernel module for a 2.6 kernel Regards Mike - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 8:09 PM Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and TTS ( text to speech)
I have done research and reading around text to speech, and was wanting to get an updated query of where everyone is with this. I have installed festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home ISO ). I also changed the directory php application to use the festival.pl to read the names of those who have not installed a greeting. It works well enough, though the only voices I can get to work ( ked and kal ) are very machine sounding. Good enough for my current purposes, but I'm having too much fun with asterisk to leave it alone at that. That said, I am asking what other's have exerienced with TTS. Do you use festival and which version -- 1.4,1.95 or the 1.96 at the asterisk at home site -- or has cepstral taken over this market? Also, I have tried to unsuccessfully install 1.95 and the 1.96. My issue with this is likely to be something i need to take up with the asterisk at home group, but is the upgrade really worth it? Any pointers or suggestions are more than welcome on learning the TTS world for linux as well as integrating with Asterisk. Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] negative timestamp error
I am setting up a new * server and everything appears to work, but I see a steady stream of these errors whenever a phone goes off hook. Google did not provide any hits. Jul 17 18:43:14 ERROR[21775]: utils.c:509 tvfix: warning negative timestamp -150096.-449125 Jul 17 18:43:14 ERROR[21775]: utils.c:509 tvfix: warning negative timestamp -150096.-429125 Jul 17 18:43:15 ERROR[21775]: utils.c:509 tvfix: warning negative timestamp -150096.-409125 Jul 17 18:43:15 ERROR[21775]: utils.c:509 tvfix: warning negative timestamp -150096.-389125 System config: CentOS 4.1 Pentium III 866 9GB SCSI \_ lvm volume 18GB SCSI/ 768MB ram MOH player is madplay - Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC BRIstuff woes
Tzafrir Cohen wrote: Hmmm... didn't you say you had a x100p card somewhere on your system? Nope - only the 2 pci HFC ISDN cards. OK - I hope I'm not buggering up the thread, but I need to answer myself as there now seems to be only 2 remaining concerns - but then I haven't yet had a chance to plug this box into an ISDN wall-socket yet - I'll get that joy tomorrow (today) afternoon!!! I'm still concerned about the auto-generated context being demo. I keep getting out-is-busy messages (remember I'm not plugged into the ISDN socket, though) Chan Extension Context Language MusicOnHold pseudodemoen 1demoen 2demoen Tzafrir, should I worry about these messages that keep coming up on the console? zaphfc: bchan rx fifo not enough bytes to receive! (z1=3015, z2=3008, wanted 8 got 7), probably a buffer overrun. (the z1= and z2= numbers keep changing - message repeats circa every minute or so) (I've fixed the registration of z1 and z2 - iax_additional.conf had the wrong phone names in it) Thanks yet again, good night, Zoltan. -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meet Me - this is not a valid conferencenumber, please try again
Just make sure asterisk can access the proper device files, run with verbosity to check that out. On 7/15/05, Erdem HAKİ [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 parport40201 2 parport_pc,lp autofs426181 0 sunrpc164485 1 zaptel208132 1 ztdummy crc_ccitt 2113 1 zaptel Thanks Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, July 15, 2005 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Meet Me - this is not a valid conferencenumber, please try again On Fri, 2005-07-15 at 16:44 +0300, Erdem HAKİ wrote: [EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 . . . Where's zaptel? modprobe ztdummy should have loaded zaptel. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kascaded call queue
Sorry for the delayed response. You should be able to get an answer for your problem by looking at this recipe: http://www.oinko.net/astrecipes/index.php?from=1q=astrecipes/ configuring+a+queue - Waldo On Jul 8, 2005, at 4:42 AM, Elmar Haneke wrote: Hi, how can I setup an callqueue wich has beside the groups of phones usually routing calls to an second group to which calls should be routed which oterwise would stay too log in queue (e.g. after 3 Minutes). Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System Jsut hangs Up
I took care of my earlier problem. But now if I call in it just says goodbye, And on my extension no matter what I do it seems to just hang up on me immediately. Its a slackware 10.1 box with Digium 22b card. I am running AMP so its mysql driven. Im not seeing any errors. It just hangs up. Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT Number of Agents for Tech Support Call Center
This is off topic but also seems like the best place to get an educated answer. I am looking for an estimated ratio of users to agents. The call center will handle all calls for the enterprise and hand off tickets to tier two and three if they cannot be resolved by the intial call. Supporting Windows systems, logins, MS apps, printers... etc, nothing special. Does anyone know the magic ratio off hand? Thanks, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours
Matt Riddell wrote: Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten. This meant that when you dialed an extension which played back these files, the video wouldn't start unless you sent video first. Now, if you don't click send video in Xten until after you hear the beep for recording, it works. It will play you the video just by dialing the extension. I'm using the latest beta of eyebeam by the way. Have you changed something for that? What is the dialplan and sip.conf for that part? a. can you call the mailbox to retrieve the video message? b. can you send the video mail via email? c. what format is the recording? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours
Ronald Wiplinger wrote: Have you changed something for that? What is the dialplan and sip.conf for that part? Nope, just standard. Don't forget you need videosupport=true Also, use h263 flat not h263+. a. can you call the mailbox to retrieve the video message? Yes. Also the announcements are as video. b. can you send the video mail via email? Seems to just send the audio. c. what format is the recording? Well it depends, but mine makes one gsm file and one h263 file. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Jsut hangs Up
Hard to say without seeing all config files. [EMAIL PROTECTED] is an easy way to get a running system with AMP. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 3:52 PM Subject: [Asterisk-Users] System Jsut hangs Up I took care of my earlier problem. But now if I call in it just says goodbye, And on my extension no matter what I do it seems to just hang up on me immediately. Its a slackware 10.1 box with Digium 22b card. I am running AMP so its mysql driven. Im not seeing any errors. It just hangs up. Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SYMBOL NETVISION II NP-3010
On 7/15/05, Barton Fisher [EMAIL PROTECTED] wrote: I was looking at these SYMBOL NETVISION II NP-3010 VoIP TCP/IP WIRELESS PHONES - I know they have been discontinued. Am I asking for trouble to buy some of these for use on Asterisk? TIA Bart Bart: I purchased some of these a while back for about $30 US and than never got motivated enough, so I can't give any pointers to configuration, except for the actual phone. They seem to be sneaky little devils on the phone for keypad configuration; one of Symbol's cable may be required (it can't be readily made: serial on one end and custom connector that no distributor seems to carry on the other). They are relatively cheap, though. They also only do H.323, so be prepared to play around with that for a bit. -Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT Number of Agents for Tech Support Call Center
HI there Try this http://www.erlang.com/calculator/erlc/ http://www.erlang.com/calculator/erlb/ Steve Totaro wrote: This is off topic but also seems like the best place to get an educated answer. I am looking for an estimated ratio of users to agents. The call center will handle all calls for the enterprise and hand off tickets to tier two and three if they cannot be resolved by the intial call. Supporting Windows systems, logins, MS apps, printers... etc, nothing special. Does anyone know the magic ratio off hand? Thanks, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP 200 and Asterisk.
I have a few UIP200s working just fine. Below is sample sip.conf from a production system as well as the corresponding uniden{MAC}.txt for tftp boot.sip.conf:[general]context=defaultport=5060bindaddr=0.0.0.0srvlookup=yesrelaxdtmf=yesrtptimeout=60rtpholdtimeout=300localnet=10.0.10.0/255.255.255.0[3211]type=friendusername=3211secret=3211host=dynamiccontext=def_INTcanreinvite=no;dtmfmode=inbandnat=never[EMAIL PROTECTED]callerid="Sample"3211disallow=allallow=ulawuniden{MAC}.txt:OverwriteLocalSettings YESProxyServer 10.0.10.13ProxyServerPort 0OutboundProxy1 10.0.10.13OutboundProxy1Port 0OutboundProxy2 0.0.0.0OutboundProxy2Port 0EmergencyProxy 0.0.0.0EmergencyProxyPort 0Registrar1 10.0.10.13Registrar1Port 0Registrar2 0.0.0.0Registrar2Port 0RegisterExpireSec 3600Q_Param 50RegisterExpireLimitPercent 10Register403RetrySec 1200SipPort 5060SRVRecordName -FailoverRetrySec 4InterDigitTimer 2500SessionTimerSupport ONSessionTimerRefresher ONSessionTimerMin 60TimerInterval0 300TimerInterval1 150G711MuTxPacketLength 20G711MuJitterBufferLength 10G711MuJitterBufferMax 200G711ATxPacketLength 20G711AJitterBufferLength 10G711AJitterBufferMax 200G729TxPacketLength 20G729JitterBufferLength 10G729JitterBufferMax 200LongHoldAlertPeriod 360RTPPortBase 25000DiffServMode OFFDefaultDiffServParam 192RTPDiffServParam 160VlanMode DISABLEVlanID 1PcVlanID 2#TftpAddress 10.0.10.13EnableSNTP yesSntpServerIP 209.132.176.4TimeZone -5EnableDST YESSntpRetrySec 1800PreferredCodec g711u,g711a,g729Language EnglishCallWaiting EnableEmergencyProxyPrefix ** StunServerAddr 0.0.0.0StunServerPort 0StunServerUpdateSec 0StunServerNATKeepAliveSec 0DirectIpDialing EnableAllowSharpAsDial YesDTMFFlashEvent DisableHotLineDisplay DisableUseCustomizedUserAgentName NoCustomizedUserAgentName Uniden^Customized^UAAdminPassword 1234/AutoFirmwareUpdate YESFirmwareFileName uip200_463enc.pacFirmwareVersion BS4.63ProgrammableKey1 OneTouchDialProgrammableKey2 OneTouchDialProgrammableKey3 OneTouchDialProgrammableKey4 CallForwardProgrammableKey5 TwoTouchDialProgrammableKey6 DoNotDisturbProgrammableKey7 VMAProgrammableKey8 MuteOneTouchKey1 18005558355OneTouchKey2 5553456OneTouchKey3 3456OneTouchKey4 3457OneTouchKey5 3458OneTouchKey6 918005558355OneTouchKey7 3459OneTouchKey8 3450TwoTouchDigit0 3469TwoTouchDigit1 3450TwoTouchDigit2 4420TwoTouchDigit3 4421TwoTouchDigit4 4422TwoTouchDigit5 4423TwoTouchDigit6 4424TwoTouchDigit7 4425TwoTouchDigit8 4426TwoTouchDigit9 4427HotLineNumber -VmaDirectCallNo 3685VmwiLampIndicator EnableTimeDisplay EnableImportPhoneBook NoMyLcdDisplay 211MyDialNumber 3211DisplayName 3211UserNameForProxy 3211PasswordForProxy 3211UserNameForRegistrar 3211PasswordForRegistrar 3211- WaldoOn Jul 11, 2005, at 2:39 PM, Heath Oderman wrote: Adam, I've tried both the [heath] heading and the [31521] heading. I figure the 31521 was right because the registration error message says [EMAIL PROTECTED] I've tried host = dynamic and defaultip = 172.x No combination of those above settings scores me a successful registration. hFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mark Edwards Sent: Tuesday, July 05, 2005 5:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uniden UIP 200 and Asterisk. Unless I'm very much mistaken you want to get rid of either the host=dynamic or the defaultip=something host=dynamic indicates the device is getting an IP from dhcp and it will tell * what it is when it registers. defaultip=something indicates that the device is staticip. Devices like this are normally dynamic so try losing the defaultip entry cheers Mark On 7/6/05, Heath Oderman [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be
Re: [Asterisk-Users] Queue Log
I'm am writing a small php program to pull some info out of our Asterisk's queue_log. I'm having trouble figuring out what some of the parameters mean. Here's an example: 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25 (716)250-3405 1st field is current date/time in unixtime format 2nd field is channels unique-id which is unixtime of when call first arrived, a . and the PID of the asterisk process which accepted the call (or something like that). 3rd field is the queue name I'm not sure what the 4th field is... value of NONE doesn't tell me much :) Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Panasonic KX-T7665 and Asterisk?
Hello, Is it possible to connect the Panasonic KX-T7665 IP phone to Asterisk? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and Asterisk@home
That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
I guess I was wondering if it was crippled in some way. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between Asterisk and Asterisk@home
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. It should be mentioned that the [EMAIL PROTECTED] CD-ROM will silently (...or with very little fanfare) blow away any partitions you may have on your PC and install Linux and [EMAIL PROTECTED] So Be very carefull is booting on the [EMAIL PROTECTED] CD-ROM on a system that you don't want to sacrifice. That said, I have several [EMAIL PROTECTED] installs and they work fine for my purposes. I have dual TDM400P's with 8 incoming PSNT (TDM) lines from the Telco, and a combination of SIP and IAX2 hard and softphones. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Not in any way that you would be worried about for at least a while. It does restrict what you can customize but by the time you get to that stage (I still haven't - you can install your own version) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Felder Sent: Sunday, 17 July 2005 11:33 PM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] I guess I was wondering if it was crippled in some way. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] not accepting IAX calls from outside
I've been banging my head with this all day. I today switched from a very old CVS build to AAH1.3 and so far everything has been easy. However I cannot accept calls from a previously working IAX trunk. I've set up an trunk with all the same credentials as before and can call the folks at the other pbx. However whenever they call me I tell them that I don't have an extension/context by the name they dialed. Any ideas? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
I think that AAH would suit you fine for a small business application. It is not crippled in any way. Far from it in fact. It has more features installed as default than the latest CVS version. It is however based on the latest official release of the Asterisk code (whatever that is at the time you download the disk) and so may not have some of the latest bug fixes etc. Go for it! I've install a few for my customers and they love it. Mark Michael Felder wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) - Playback , MP3Player and Musiconhold not working
I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise - extensionns.conf - [general] static=yes writeprotect=no [sip] exten = 123,1,Dial(SIP/123,20,tr) exten = 123,2,Voicemail,u123 exten = 123,102,Voicemail,b123 exten = 321,1,Dial(SIP/321,20,tr) exten = 321,2,Voicemail,u321 exten = 321,102,Voicemail,b321 exten = 100,1,Answer ;exten = 100,2,MusicOnHold(default) ;exten = 100,2,Playback(tt-weasels,skip) exten = 100,2,MP3Player(/usr/local/share/asterisk/mohmp3/fpm-sunshine.mp3) ;exten = 100,3,Voicemail(100) exten = 100,4,Hangup - I see the mpg123 running /usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s fpm-calm-river.mp3 same happens if I use Playback or MusiconHold. ( I hear noise ) I'm confused, I load the module of the sound card to the kernel.. (this is related to this problem ?? ) + FreeBSD Audio Driver (newpcm) Installed devices: pcm0: VIA VT82C686A at io 0xcc00 irq 10 kld snd_via82c686 (1p/1r/0v channels duplex default) + Any idea or help, will be apreciated. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Michael Felder wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike [EMAIL PROTECTED] is a Linux distribution that makes it easy to install Asterisk and a few other Asterisk related applications (like AMP). Asterisk is the main application that makes [EMAIL PROTECTED] different from the CentOS distro (which is what it is based off of). Your question is a little like what is the difference between Apache and Fedora (Fedora usually includes Apache on server installs). That being said, [EMAIL PROTECTED] is crippled, but only because by default is uses AMP, which somewhat limits what you can change via it's interface. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Mark Phillips wrote: I think that AAH would suit you fine for a small business application. It is not crippled in any way. Far from it in fact. It has more features installed as default than the latest CVS version. Not to sound rude, but that doesn't make any sense. Or I'm reading it wrong :). [EMAIL PROTECTED] is a distro that by default includes a release version of Asterisk STABLE. First of all, you can't even directly compare the two (Asterisk vs. [EMAIL PROTECTED]). Would you compare an engine by itself to an entire car? No, you wouldn't. Don't do it here, either. Secondly, unless they are doing some mad branching and patching, there is NO WAY that the version of stable that they include has more Asterisk features than CVS HEAD. They include some applications that help enable some Asterisk features (mysql, mpg123, etc), but there is no reason that you can't get all of that with another distro (albeit not as easily) and Asterisk stable, or even more Asterisk features with all of those applications and CVS HEAD. It is however based on the latest official release of the Asterisk code (whatever that is at the time you download the disk) and so may not have some of the latest bug fixes etc. Go for it! I've install a few for my customers and they love it. Great! Use what works! Mark -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC BRIstuff woes
On Mon, Jul 18, 2005 at 12:52:23AM +0200, Zoltan Szecsei wrote: Tzafrir Cohen wrote: Hmmm... didn't you say you had a x100p card somewhere on your system? Nope - only the 2 pci HFC ISDN cards. OK - I hope I'm not buggering up the thread, but I need to answer myself as there now seems to be only 2 remaining concerns - but then I haven't yet had a chance to plug this box into an ISDN wall-socket yet - I'll get that joy tomorrow (today) afternoon!!! I'm still concerned about the auto-generated context being demo. So edit zapata*.conf and set the 'context=' line to the context of your choosing. As far as chan_zap is concerened, this is really just an arbitrary string. I keep getting out-is-busy messages (remember I'm not plugged into the ISDN socket, though) Chan Extension Context Language MusicOnHold pseudodemoen 1demoen 2demoen Tzafrir, should I worry about these messages that keep coming up on the console? zaphfc: bchan rx fifo not enough bytes to receive! (z1=3015, z2=3008, wanted 8 got 7), probably a buffer overrun. (the z1= and z2= numbers keep changing - message repeats circa every minute or so) Sorry, no idea. But you still seem to use just one card. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * CVS-HEAD and ASTCC Intermittent issue
Hie! I've installed Asterisk CVS-HEAD with ASTCC. The problem i'm facing is that the astcc.agi script completes when the recipient picks up the call. When the astcc.agi completes is returns 0 bill time but both end still able to talk. It occurs intermittently, any one facing the same issue? Asterisk Console - == Spawn extension (sip, 77, 2) exited non-zero on 'SIP/1112-15a3' -- Executing Answer(SIP/1112-9696, ) in new stack -- Executing DeadAGI(SIP/1112-9696, astcc.agi|1112|) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi -- Playing 'digits/4' (language 'en') -- Playing 'astcc-pin' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/40' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/20' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/20' (language 'en') -- Playing 'digits/3' (language 'en') -- AGI Script Executing Application: (DIAL) Options: (Local/[EMAIL PROTECTED]|30|HL/n(78:6:3)) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/|30|tr) in new stack -- Called [EMAIL PROTECTED] -- Called -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/-0c16 is ringing -- SIP/-0c16 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered SIP/1112-9696 -- AGI Script astcc.agi completed, returning 0 == Spawn extension (sip, , 1) exited non-zero on 'SIP/1112-9696' Regards See Hoe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Jsut hangs Up
On Sun, Jul 17, 2005 at 09:52:35PM -0400, Tim King wrote: I took care of my earlier problem. But now if I call in it just says goodbye, And on my extension no matter what I do it seems to just hang up on me immediately. It's a slackware 10.1 box with Digium 22b card. I am running AMP so its mysql driven. I'm not seeing any errors. It just hangs up. Do you have a shell on that system? Can you see what is asterisk doing? Is it hung on 100%CPU? in a D state? If you run asterisk with real-time priority (-p): try removing it so that if asterisk takes 100% CPU it won't block the whole system. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem while capturing DTMF digits in AGI
Hi All,
I have some problem while capturing DTMF digits in AGI
script.
My configuration for user is ..
[9009]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
[9010]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
And the script read like ...
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $tests = 0;
my $pass = 0;
my $fail = 0;
#setup callback
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
my $timeout = 3000; # 3 second timeout
my $wait = 1; # true
my $outgoing_number;
while($wait) {
my $digit = $AGI-wait_for_digit($timeout);
print STDERR Digit ($digit)\n;
if ($digit 0) {
$digit -= 48;
$outgoing_number .= $digit;
}
else { #stop waiting for more digits
$wait = 0;
}
}
print STDERR Outgoing number ($outgoing_number)\n;
--
I will get the output as Outgoing number () and
digits(0)
Am I missing something here? Please do help me in this
regard.
Regards
Somesh S Shanbhag
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Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote: On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. If you wanted a knoppix install there is knopsterix, which can save your config without overwriting your whole drive without prompting you to save a partition or two. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
trixter http://www.0xdecafbad.com wrote: On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote: On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. If you wanted a knoppix install there is knopsterix, which can save your config without overwriting your whole drive without prompting you to save a partition or two. I should be releasing a much improved Live version of AstLinux within a week or so. A test version was announced on my mailing list a while ago, with pretty good results so far. It will be AstLinux 0.2.8, and available as an ISO (as well as the Windows install package, disk images, etc.) I'll let everyone here know when it's released. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem while capturing DTMF digits in AGI
Hello,
You must answer the channel
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $tests = 0;
my $pass = 0;
my $fail = 0;
#setup callback
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
$AGI-answer();#---Answer the channel
Cheers,
Madhawa
somesh s wrote:
Hi All,
I have some problem while capturing DTMF digits in AGI
script.
My configuration for user is ..
[9009]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
[9010]
type=friend
host=dynamic
context=default
dtmfmode=rfc2833
dtmfrelax=no
disallow=all
allow=ulaw
allow=h263
canreinvite=yes
And the script read like ...
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $tests = 0;
my $pass = 0;
my $fail = 0;
#setup callback
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
my $timeout = 3000; # 3 second timeout
my $wait = 1; # true
my $outgoing_number;
while($wait) {
my $digit = $AGI-wait_for_digit($timeout);
print STDERR Digit ($digit)\n;
if ($digit 0) {
$digit -= 48;
$outgoing_number .= $digit;
}
else { #stop waiting for more digits
$wait = 0;
}
}
print STDERR Outgoing number ($outgoing_number)\n;
--
I will get the output as Outgoing number () and
digits(0)
Am I missing something here? Please do help me in this
regard.
Regards
Somesh S Shanbhag
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[Asterisk-Users] Debugging Realtime Asterisk
Is there any way to get debug info on res_odbc? I get the following but this is the last I ever see of anything ODBC related. Obviously, my extensions are not working from the database, but I can connect to ODBC via isql and run queries just fine. Jul 17 22:12:13 NOTICE[3923]: res_odbc.c:215 load_odbc_config: registered database handle 'asterisk' dsn-[asterisk] Jul 17 22:12:13 NOTICE[3923]: res_odbc.c:473 odbc_obj_connect: Connecting asterisk Jul 17 22:12:14 NOTICE[3923]: res_odbc.c:488 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] Jul 17 22:12:14 NOTICE[3923]: res_odbc.c:518 load_module: res_odbc loaded. [res_config_odbc.so] = (ODBC Configuration) Jul 17 22:12:14 NOTICE[3923]: config.c:836 ast_config_engine_register: Registered Config Engine odbc __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe wcfxo fails.
Hello Mike, Is your Asterisk box running for a HAM's project that I am working on ? 73, F6HQZ, Francois BERGERET. - Original Message - From: Michael J. Tubby G8TIC To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, July 18, 2005 12:30 AM Subject: Re: [Asterisk-Users] modprobe wcfxo fails. Tim, Yes, it was resolved but cannot remember how... I'm using RedHat FC3, Asterisk 1.0.9 and it all just works assuming that you: - RTFM with respect to udev setup - use "make linux26" to build the kernel module for a 2.6 kernel Regards Mike - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 8:09 PM Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
