Re: [asterisk-users] Realtime SIP, multiple AX servers question
3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot: Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as regseconds, lastms, ipadr, etc. However, with some ATA's we are experiencing a problem as follows: 1. ATA reaches its re-registration timeout, which we typically configure to be 60 minutes. 2. ATA re-queries DNS SRV record, and ends up re-registering with a different AX server than it was on previously. 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc). 4. The old AX server, after a few more minutes, notices that the ATA has vanished, and therefore clears out these same fields. Oh, that's an interesting observation. Depending on how you see it, it's a bug or a feature request. Code-wise what you could do is that Asterisk could retrieve the information from realtime. If the sysname is not the same as the systems, it let the information be. If it's the same sysname, then erase the information. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Forwarding
- Original Message - --[ UxBoD ]-- wrote: - Original Message - Yes exactly that indeed. Though Asterisk appears to ignore which context the user is in and selects default instead. Beginning to think that it is a bug. I got it figured out. In your voicemail.conf, search for the option searchcontexts=yes And enable it. Doug Sorry for the late reply! While that does allow it to work it is not appropriate in a multi-tenant environment where the same extension could exist in different contexts. Will file a bug for this and the configuration we are using looks correct. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Go from CALLINGout to just CALLING
Hello list,
how can I go from CALLINGout to just CALLING ?
I've tried :
exten = s,n,Set(newVAR=${CUT(CALLINGout,,3)})
or
exten = s,n,Set(newVAR=$[CUT(CALLINGout,,3)])
But no result :
[Jan 4 11:10:12] -- Executing [...@from-s:34] NoOp(SIP/s2-003b,
newVAR=) in new stack
Asterisk 1.6.10 here.
Kind regards,
Jonas.
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Re: [asterisk-users] Log and forward calls to cellphone?
On Sat, 01 Jan 2011 23:32:15 +, Sebastian s...@open-t.co.uk wrote: Anyway - there is a third option - which I have been using with some success. I connected my softphone on my laptop to my Asterisk server at home (through OpenVPN for extra security - but this is not compulsory). [...] As a last alternative - a slight improvement on the above. If you can get a smartphone with Android - which would let you run SIP over 3G - you should have true free voice divert. Thanks Sebastian for the tip. The goal is to 1) have clients call the usual landline number instead of asking them to try a cellphone in case no one's home, 2) get Asterisk to handle the call, 3) have the cellphone ring with the CID of the original caller instead of Asterisk's. It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. Does someone know... 1. how reliable 3G Internet access is in Europe in cities? 2. what smartphone supports installing an SIP + OpenVPN clients? 3. how much juice those things need to keep those applications + 3G connection running for hours each day? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question About Conferencing Capabilities
Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation. regards dhaval On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton siobhan.plugge...@gmail.com wrote: My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
Le 03/01/2011 18:28, Gilles a écrit : On Mon, 03 Jan 2011 12:27:56 +0100, Administrator TOOTAI ad...@tootai.net wrote: As you are a Free Telecom customer, why not using your freephonie account to forward incoming calls to your mobile? Thanks for the tip, but experience shows that their SIP access sucks (not reliable, quality NOK). That's why I got a VOSP account. Don't know the meaning of VOSP but you can do it with any SIP/IAX/H323/... provider. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Go from CALLINGout to just CALLING
On Tue, Jan 4, 2011 at 5:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
how can I go from CALLINGout to just CALLING ?
I've tried :
exten = s,n,Set(newVAR=${CUT(CALLINGout,,3)})
or
exten = s,n,Set(newVAR=$[CUT(CALLINGout,,3)])
But no result :
[Jan 4 11:10:12] -- Executing [...@from-s:34] NoOp(SIP/s2-003b,
newVAR=) in new stack
Asterisk 1.6.10 here.
I don't think CUT does what you think it does.
When using CUT, the second argument should be a delimiter, (hyphen,
pipe, comma, etc.)
I can't really tell what you are trying to achieve, but if CALLINGout
is the value
of a variable, say X, and you want just the first 6 characters, you
could use (maybe):
exten = s,n,Set(newVAR=${X:0:6})
HTH,
-Bob
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Re: [asterisk-users] VoIP PoE phones for restaurant
On 01/03/2011 07:53 PM, cjwstudios wrote: Andy, The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the 2626 can support vlans you can isolate data and voice. Make sure to spec a UPS on the PoE switch. CJW, Awesome. Thanks for the input. For some reason or another I figured EOL wasn't such a bad thing as I could pick up the phones for cheap on ebay or something; but maybe this isn't the best of plans. The IP335 is on average about $10 more than the 501 or 320 new anyway. I thought that the 2610-24/12-PWR had the ability for VLAN as well? Not that it matters, it looks like I can get the 2626-PWR for under $600, and that fills out POE to all the ports. Is it possible that I can run one cable to the phone, then run a cable from the phone to a computer or another device and have those the phone and computer or other device be on separate networks? I'm sorry if this sounds newbish; I'm still learning. -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
On Tuesday 04 Jan 2011, Gilles wrote: Thanks Sebastian for the tip. The goal is to 1) have clients call the usual landline number instead of asking them to try a cellphone in case no one's home, 2) get Asterisk to handle the call, 3) have the cellphone ring with the CID of the original caller instead of Asterisk's. The problem with doing no. 3 is, if you are routing the call over the PSTN at any rate, your telephone company will (silently) *drop* the caller ID if the number you are presenting does not actually belong to you. This is *good* most of the time, because it means you can trust other people's caller ID to be accurate (and untrustworthy caller ID makes caller ID pointless). We first met this when we ordered our second E1 line and batch of presentation numbers. As a result of a mistake on somebody's part, the two lines appeared (according to BT's records) to belong to different companies. As a result, approximately half our calls were going out anonymously; because if we were trying to go out on span 2 but using a number that was only allowed on span 1, or vice versa, then the ident would get stripped somewhere along the way. Diagnosing this obscure fault rather stretched the definition of fun :/ -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/03/2011 07:08 PM, Steve Underwood wrote: On 01/04/2011 04:22 AM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: I'll try. 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. That's not true. Modern echo cancellers normally disable completely. It is arguable whether they should disable completely for FAX, but they need to behave properly for all modems. For any duplex modem, disabling only the NLP is useless. They need to cancel end to end, so they don't get upset by a continuously adapting canceller, and so they can minimise the issues caused by the highly non-linear G.711 channel. This doesn't match up with what the manufacturers of the two G.168 ECs that Digium distributes have told me personally about their products. Their ECs behave differently for FAX and 'regular' modems, but they do that based on the detection of the V.21 preamble, ANSam and other signals in addition to CED, which seemed to be much more detail than was warranted in my response to the OP :-) 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. Also, do Digium cards with HW Echo Cancellation detect the CNG tones in hardware? If so, how does the faxdetect setting in DAHDI affect that behavior? No, none of the Digium HW ECs detect and report CNG tones via the DSP; CNG tone detection is still done on the host CPU. 'faxdetect' is not set in DAHDI, it's set in chan_dahdi.conf. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/03/2011 06:47 PM, Thomas Rymes wrote: On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: [snip] 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. OK. Either way, though, the changes to echo cancellation are not affected by the faxdetect setting, right? That is correct; the faxdetect setting and the echo canceller behavior are completely unrelated. 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. Great. This is the typical usage, I presume, directing fax machines to FFA, Hylafax, another fax machine, or hangup (if this isn't a fax line). Is there a time limit to when DAHDI listens for faxes (say the first 10 seconds of a call?), or might it detect one in the middle of a ten minute call? I haven't double-checked, but I believe the software DSP will be in place on the call until it sees a CNG tone, regardless of when that happens during the call. 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. [snip] I figured that must be it. Presumedly you might use this to perform some activity on an outgoing fax prior to sending it, such as logging something, etc? Maybe send it to FFA, receive it, and e-mail it to another server that faxes it out on a local number to save toll calls, etc? Thanks for the clarification, there's a lot of conflicting info out there. Feel free to comment on wiki.asterisk.org if any of the information there led you astray; we'd like to get that to be the most accurate place for people to find this sort of information. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP, multiple AX servers question
Thanks Olle. Do you suppose I am the first Asterisk user to discover this behavior? I would find that hard to believe that I'm the first person to notice... Your idea for how to deal with sounds reasonable.. Thank you, Bryan On Jan 4, 2011, at 12:18 AM, Olle E. Johansson wrote: 3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot: Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as regseconds, lastms, ipadr, etc. However, with some ATA's we are experiencing a problem as follows: 1. ATA reaches its re-registration timeout, which we typically configure to be 60 minutes. 2. ATA re-queries DNS SRV record, and ends up re-registering with a different AX server than it was on previously. 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc). 4. The old AX server, after a few more minutes, notices that the ATA has vanished, and therefore clears out these same fields. Oh, that's an interesting observation. Depending on how you see it, it's a bug or a feature request. Code-wise what you could do is that Asterisk could retrieve the information from realtime. If the sysname is not the same as the systems, it let the information be. If it's the same sysname, then erase the information. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forwrading but call transfer back
Hi All, I have weird requirement for call forwarding. I have forward all call from A to B extension because A is very busy and sometime not available so B is taking care of all forwarding call from A. but in some case B need to transfer call to A and in this case call coming back to B again because of forwarding enabled. How to get rid on this condition ? How could B can transfer call to A ? Thanks, Satish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Contact center - How to Gmail-like label to incoming email
Hi, Though this has no direct relation with Asterisk, I think Asterisk users in Contact Centers might have an interesting answer. Using Gmail, it's rather easy to label an incoming email so that any related email (reply) inherits this label and are commonly displayed together in threads. Using Google, I found this related RFC5256 ( http://tools.ietf.org/search/rfc5256). Which (preferably Open Source) scriptable mail engine would support such Thread-labelling feature so that incoming emails could be appropriately be marked and routed to the right agents, depending on simple rules such as : al...@example.org is marked as VIP in CRM and emails from VIP are to be labelled as VIP. Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues, priorities and (miscalculated) holdtimes
Anyone ever noticed that the reported holdtime is wrong when there are different priorities? Also talktime is 0, but for the moment I don't care. queue show test reports: test has 23 calls (max unlimited) in 'ringall' strategy (193s holdtime, 0s talktime) [...] Callers: 1. Local/3...@default-8828;2 (wait: 3:32, prio: 15) 2. Local/3...@default-8361;2 (wait: 3:32, prio: 15) [...] 6. Local/1...@default-b575;2 (wait: 9:25, prio: 10) 7. Local/1...@default-fce5;2 (wait: 9:21, prio: 10) [...] 22. Local/3...@default-89fb;2 (wait: 6:45, prio: 5) 23. Local/3...@default-7264;2 (wait: 0:08, prio: 5) The reported holdtime to the caller is 3 minutes! The wait time of caller number 1, instead of the 9 minutes holdtime of caller number 6. This is a realtime Queue on 1.6.2.13. Also can the priority be changed dynamically (using AMI)? For now I'm using: exten = s,n,Set(QUEUE_PRIO=5) exten = s,n,Queue(test,twrC,,,900) exten = s,n,Set(QUEUE_PRIO=15) exten = s,n,Queue(test,twr,,,3600) some callers and exten = s,n,Set(QUEUE_PRIO=10) exten = s,n,Queue(test,twr,,,3600) for all others. Changing the the priority dynamically depending on context and waittime should avoid the wrong reported holdtime. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH problems (asterisk 1.4.38)
Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory with the default musiconhold.conf file. When I change musiconhold.conf to this: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 (and have converted all the wav files to mp3 and put them in /var/lib/asterisk/nohmp3) 'moh reload' works fine and so does 'moh show classes' but 'moh show files' does not show any files (even though the .mp3 files are in that directory) and of course MOH still does not work I am not sure how to 'debug' from this point. Any help would be greatly appreciated! thanks earl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
Hi, On 01/04/2011 10:50 AM, Gilles wrote: On Sat, 01 Jan 2011 23:32:15 +, Sebastians...@open-t.co.uk wrote: Anyway - there is a third option - which I have been using with some success. I connected my softphone on my laptop to my Asterisk server at home (through OpenVPN for extra security - but this is not compulsory). [...] As a last alternative - a slight improvement on the above. If you can get a smartphone with Android - which would let you run SIP over 3G - you should have true free voice divert. Thanks Sebastian for the tip. The goal is to 1) have clients call the usual landline number instead of asking them to try a cellphone in case no one's home, 2) get Asterisk to handle the call, 3) have the cellphone ring with the CID of the original caller instead of Asterisk's. It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. Does someone know... 1. how reliable 3G Internet access is in Europe in cities? I can only speak for the UK. In the UK - Three seems to be one of the best providers (in my experience). However, coverage quality varies throughout the country, and I have clients on O2, T-Mobile and Vodafone - with varying results. It is, by its very nature, a connection which will vary continually in bandwidth and reliability with the time and location. 2. what smartphone supports installing an SIP + OpenVPN clients? Looking around, it seems to me that any Android phone should be able to have SIP clients installed. If anybody knows of any manufacturer or operator imposed blocks - I would love to know. One of the more popular SIP clients (www.sipdroid.org) doesn't seem to mention any possible impediments to installing it on any Android phone (1.5 and above) 3. how much juice those things need to keep those applications + 3G connection running for hours each day? Again, at least according to www.sipdroid.org FAQ - it seems that it shouldn't make any extra difference. I suppose it depends on the battery size. They claim a 3 days standby - but don't say which phone did they test it on. They also claim that a stock Asterisk talking to a SIP client on Android is not ideal in terms of battery life for the Android phone - but I really can't think why. If anybody here has some ideas - would be great. One other thing to watch out for is operator imposed contractual restrictions. Many mobile/3G operators expressly forbid running any type of VoIP through their network in the contract (you can still use the phone + SIP over wifi, though). However, I believe that if you run it through OpenVPN - they shouldn't be able to tell. Again, if anybody has any info on this, or knows otherwise - I would love to know. One of the openvpn implementations for Android is TunnelDroid (http://sourceforge.net/projects/tunneldroid/). This one needs the phone to be rooted - so when searching for a phone - make sure it has a (hopefully easy) rooting procedure. I don't know if there is an openvpn implementation for Android which doesn't need the phone to be rooted - but considering you need extra kernel modules (the tun device) I would have thought rooting is essential. Sorry to keep on butting in. I've been interested in SIP on Android for a while now - so this just gave me more incentives to actually do the research :-) Sebastian Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/04/2011 8:55 AM, Kevin P. Fleming wrote: On 01/03/2011 06:47 PM, Thomas Rymes wrote: On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: [snip] OK. Either way, though, the changes to echo cancellation are not affected by the faxdetect setting, right? That is correct; the faxdetect setting and the echo canceller behavior are completely unrelated. Excellent. [snip] Is there a time limit to when DAHDI listens for faxes (say the first 10 seconds of a call?), or might it detect one in the middle of a ten minute call? I haven't double-checked, but I believe the software DSP will be in place on the call until it sees a CNG tone, regardless of when that happens during the call. Wouldn't it make sense to be able to specify a time period after which chan_dahdi disables fax detection? Only calls that begin with a voice call and end with a fax would benefit from detection after the initial ~8 seconds of a call, unless I am overlooking something. If the DSP keeps listening and detects a spurious fax tone (I know I have seen the human voice incorrectly identified as CNG), it will send the call off to the fax extension if one exists in the same context. In fact, we ran into some issues with exactly that happening. [snip] Thanks for the clarification, there's a lot of conflicting info out there. Feel free to comment on wiki.asterisk.org if any of the information there led you astray; we'd like to get that to be the most accurate place for people to find this sort of information. I'll give it a look. I had not specifically looked at the asterisk wiki, but Google searches brought up lots of messages confusing the fax operation of the echo canceler with the faxdetect= setting for DAHDI/Zaptel. Thanks again, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant
On Tue, Jan 4, 2011 at 8:52 AM, Andy Graybeal andy.grayb...@casanueva.com wrote: On 01/03/2011 07:53 PM, cjwstudios wrote: Andy, The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the 2626 can support vlans you can isolate data and voice. Make sure to spec a UPS on the PoE switch. CJW, Awesome. Thanks for the input. For some reason or another I figured EOL wasn't such a bad thing as I could pick up the phones for cheap on ebay or something; but maybe this isn't the best of plans. The IP335 is on average about $10 more than the 501 or 320 new anyway. I thought that the 2610-24/12-PWR had the ability for VLAN as well? Not that it matters, it looks like I can get the 2626-PWR for under $600, and that fills out POE to all the ports. Is it possible that I can run one cable to the phone, then run a cable from the phone to a computer or another device and have those the phone and computer or other device be on separate networks? I'm sorry if this sounds newbish; I'm still learning. The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question About Conferencing Capabilities
Anyone else know about the holding concurrent conferences (and switching back and forth) issue ? Is it possible? And can you set up dynamic conferences that continue even when the initiator leaves? Thanks! On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation. regards dhaval On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton siobhan.plugge...@gmail.com wrote: My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
On 01/04/2011 01:55 PM, A J Stiles wrote: On Tuesday 04 Jan 2011, Gilles wrote: Thanks Sebastian for the tip. The goal is to 1) have clients call the usual landline number instead of asking them to try a cellphone in case no one's home, 2) get Asterisk to handle the call, 3) have the cellphone ring with the CID of the original caller instead of Asterisk's. The problem with doing no. 3 is, if you are routing the call over the PSTN at any rate, your telephone company will (silently) *drop* the caller ID if the number you are presenting does not actually belong to you. This is *good* most of the time, because it means you can trust other people's caller ID to be accurate (and untrustworthy caller ID makes caller ID pointless). I agree with your point. That is why routing the divert part of the call through an (effectively) internal SIP extension - which is the case if you call your laptop or Android phone through SIP as an internal extension to your Asterisk server (through OpenVPN as well, optionally) has the advantage that you can transmit/present whatever Caller ID you want. Sebastian We first met this when we ordered our second E1 line and batch of presentation numbers. As a result of a mistake on somebody's part, the two lines appeared (according to BT's records) to belong to different companies. As a result, approximately half our calls were going out anonymously; because if we were trying to go out on span 2 but using a number that was only allowed on span 1, or vice versa, then the ident would get stripped somewhere along the way. Diagnosing this obscure fault rather stretched the definition of fun :/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forwrading but call transfer back
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Tuesday, January 04, 2011 9:36 AM To: asterisk-users Subject: [asterisk-users] Call forwrading but call transfer back Hi All, I have weird requirement for call forwarding. I have forward all call from A to B extension because A is very busy and sometime not available so B is taking care of all forwarding call from A. but in some case B need to transfer call to A and in this case call coming back to B again because of forwarding enabled. How to get rid on this condition ? How could B can transfer call to A ? Thanks, Satish This is a job for ex-girlfriend logic. Set up your dialplan like this (A=1001, B=1002) Exten = 1001,verbose(extension A-1001 handling) Exten = 1001,n,dial(SIP/1002) Exten = 1001/1002,n,dial(SIP/1001) If you dial 1001 from anywhere except 1002, you get sent to 1002. If you dial 1001 from 1002, you get sent to 1001. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Announce: telepathy-ring 2.1.1
This should probably go to the dev list but I think developers and users both need to see the progress. A year or two, I posted about ofono right at their launch. It seems they have come a long way and Digium or other VoIP platforms may want to make alliances with ofono. It is not a joke project or vaporware. It is backed by Nokia and Intel. http://ofono.org/ It looks like with a little glue, chan_ofono could quickly bring real cell phone capabilities to Asterisk or other voip platforms. Just an FYI, Steve Totaro -- Forwarded message -- From: Pekka Pessi ppe...@gmail.com Date: Tue, Jan 4, 2011 at 2:29 PM Subject: Announce: telepathy-ring 2.1.1 To: telepa...@lists.freedesktop.org, of...@ofono.org Telepathy-Ring 2.1.1 (too little butter over too much bread) is now available for download from: http://telepathy.freedesktop.org/releases/telepathy-ring/telepathy-ring-2.1.1.tar.gz md5sum: f2ae9dd104cc16eec548884beead85a7 telepathy-ring-2.1.1.tar.gz sha1sum: f46bb30dda9ba40a057c128af2a27c23a6a937af telepathy-ring-2.1.1.tar.gz What is it? === Telepathy-Ring a 3GPP (GSM and 3G UMTS) connection manager for Telepathy framework using oFono. It supports voice calls and short messages. What's New? === Initial conference call support has been added. The ofono interface code in modem subdirectory has been improved. The following bugs has been fixed: fd.o #32718: return InvalidHandle if calling to self or anonymous handle fd.o #30954: advertize Message mixin immutable properties fd.o #31726: use TpBaseChannel fd.o #31664: avoid deprecated tp_get_bus -- Pekka.Pessi mail at nokia.com ___ ofono mailing list of...@ofono.org http://lists.ofono.org/listinfo/ofono -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
Hi, On 01/04/2011 03:24 PM, Administrator TOOTAI wrote: Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery Any chance you could provide few more details please? Mainly which phone, what version of Android, and how many hours on standby when using OpenVPN. Also, which application were you running through OpenVPN and was it in constant use (the app). I am investigating using OpenVPN with Android - and I would find the above detail very useful. Many thanks, Sebastian [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP, multiple AX servers question
On Tuesday 04 January 2011 09:40:56 Bryan Field-Elliot wrote: Thanks Olle. Do you suppose I am the first Asterisk user to discover this behavior? I would find that hard to believe that I'm the first person to notice... It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
Hi CLI module unload res_musiconhold.so CLI module load res_musiconhold.so or service asterisk restart Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant
On 01/04/2011 8:52 AM, Andy Graybeal wrote: Is it possible that I can run one cable to the phone, then run a cable from the phone to a computer or another device and have those the phone and computer or other device be on separate networks? I'm sorry if this sounds newbish; I'm still learning. I'm no networking expert, but no one else has answered, so I'll give it a shot. It is indeed possible (quite common, actually) to run the wiring as you describe. If you want to keep the data and voice traffic separate, you can use VLANs to do so. Your switches will need to support VLANS, and you will need to configure VLANs to separate the voice and data traffic. As I understand it, though, you are still subject to the bandwidth limitations of the underlying network, so it's still possible that heavy traffic from the PC might affect the voice traffic. QOS or other methods might be used to help avoid this. For this reason, I personally prefer to keep my voice and data LANs physically separated when possible. Obviously, cost and complexity do increase somewhat. It's probably not a good solution for everyone, but it sounds like you have a pretty small installation and you might decide that the additional cost is justified. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
On 01/04/2011 12:31 PM, Earl Terwilliger wrote: Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory [snip] Have you installed mpg123 or some other program to handle the mp3 files? I am fairly certain that Asterisk cannot handle mp3 natively (most likely for licensing reasons). Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI and dialdebounce
According to https://issues.asterisk.org/view.php?id=16339 , the default value for the dialdebounce parameter of the wctdm module has been changed to 32 and is now user configurable. I have two questions: 1.) Am I correct in presuming that, if the default of 32 does not work for me, I would specify this option in the file /etc/modprobe.d/dahdi.conf (at least for my distro, which is Elastix built on CentOS)? 2.) In the issue linked above, Tzafrir asked if this value should also be changed for the wctdm24xxp module, but there is no indication as to whether the change is needed for wctdm24xxp, or if it has already been made. Can anyone clarify? Many thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing digital pri card
On 01/03/2011 9:46 PM, Matt Watson wrote: I don't imagine this would be too complicated - don't have any experience with AsteriskNOW - but on a 'vanilla' linux distro it would just be a matter of making sure dahdi is loading the correct drivers and doing a couple of minor config file updates. On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis tda...@zulily.com mailto:tda...@zulily.com wrote: I need to replace our current 1 port pri card with a quad port card. I'm currently using the newest AsteriskNOW distro. Are there any issues I should expect to run into? I'm hoping the transition will be smooth, however I havent had to do this in the past. It should be reasonably easy, but you will need to update your DAHDI configs, including chan_dahdi.conf. I think that Digium developed and included a DAHDI configuration module for FreePBX that makes that more intuitive. If you use a non-Digium card, you'll need to update those configurations, too. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone would be best being used as an intercom in a busy kitchen. I asked this some months ago; but this time around I'm writing it into this years budget. I see the 335 has HD Voice and the 321 has Clarity by Polycom. Which would be best in a noisy kitchen using the devices speaker phone? Should I seek another device for the kitchen all-together? -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems inserting dahdi modules using Debian Leni
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do not disturbe
Hi all,
I am trying to set up DND in my asterisk, I am using the following context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten =
*11,2,GotoIf($[${DND} = YES]?*11,3:*11,101)exten =
*11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten = *11,4,Playback(beep)exten =
*11,5,Hangup()exten = *11,101,Set(DB(ddisturbe/${CALLERIDNUM})=YES)exten =
*11,102,Playback(beep)exten = *11,103,Hangup() I am testing with a softphone
and when I dial *11, I receive the following log from cli:
Executing [...@a2billing:1] Set(SIP/2015-0187, DND=YES) in new stack
-- Executing [...@a2billing:2] GotoIf(SIP/2015-0187, 1?*11,3:*11,101)
in new stack-- Goto (a2billing,*11,3)-- Executing [...@a2billing:3]
Set(SIP/2015-0187, DB(ddisturbe/)=NO) in new stack-- Executing
[...@a2billing:4] Playback(SIP/2015-0187, beep) in new stack--
SIP/2015-0187 Playing 'beep.gsm' (language 'en')-- Executing
[...@a2billing:5] Hangup(SIP/2015-0187, ) in new stack == Spawn
extension (a2billing, *11, 5) exited non-zero on 'SIP/2015-0187'
Therefore, the facilite is not working!!What I am doing wrong, could somebody
point me out please?!!
Thanks in advanced!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/05/2011 02:39 AM, Tom Rymes wrote: On 01/04/2011 8:55 AM, Kevin P. Fleming wrote: On 01/03/2011 06:47 PM, Thomas Rymes wrote: On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: [snip] OK. Either way, though, the changes to echo cancellation are not affected by the faxdetect setting, right? That is correct; the faxdetect setting and the echo canceller behavior are completely unrelated. Excellent. [snip] Is there a time limit to when DAHDI listens for faxes (say the first 10 seconds of a call?), or might it detect one in the middle of a ten minute call? I haven't double-checked, but I believe the software DSP will be in place on the call until it sees a CNG tone, regardless of when that happens during the call. Wouldn't it make sense to be able to specify a time period after which chan_dahdi disables fax detection? Only calls that begin with a voice call and end with a fax would benefit from detection after the initial ~8 seconds of a call, unless I am overlooking something. If the DSP keeps listening and detects a spurious fax tone (I know I have seen the human voice incorrectly identified as CNG), it will send the call off to the fax extension if one exists in the same context. In fact, we ran into some issues with exactly that happening. It is very normal for many people to chat and then start their FAX machines, especially domestic FAX users with a FAX machine attached to their home land line. If you don't care about those your proposal is OK, otherwise. There is no excuse for false detection of FAX tone. It takes a very poor detector to mistake voice for FAX, unless the person is specifically trying to whistle the right tones (which some people are quite good at). [snip] Thanks for the clarification, there's a lot of conflicting info out there. Feel free to comment on wiki.asterisk.org if any of the information there led you astray; we'd like to get that to be the most accurate place for people to find this sort of information. I'll give it a look. I had not specifically looked at the asterisk wiki, but Google searches brought up lots of messages confusing the fax operation of the echo canceler with the faxdetect= setting for DAHDI/Zaptel. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/04/2011 09:53 PM, Kevin P. Fleming wrote: On 01/03/2011 07:08 PM, Steve Underwood wrote: On 01/04/2011 04:22 AM, Kevin P. Fleming wrote: No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. That's not true. Modern echo cancellers normally disable completely. It is arguable whether they should disable completely for FAX, but they need to behave properly for all modems. For any duplex modem, disabling only the NLP is useless. They need to cancel end to end, so they don't get upset by a continuously adapting canceller, and so they can minimise the issues caused by the highly non-linear G.711 channel. This doesn't match up with what the manufacturers of the two G.168 ECs that Digium distributes have told me personally about their products. Their ECs behave differently for FAX and 'regular' modems, but they do that based on the detection of the V.21 preamble, ANSam and other signals in addition to CED, which seemed to be much more detail than was warranted in my response to the OP :-) Well, that makes a bit more sense, but I am very skeptical about this. The Octasic canceller is highly problematic with various modems and tones, so they aren't exactly a reference model for how to do things. Reports I here of the other canceller are much more positive. Its obvious why they want to keep the canceller alive. Long echoes over VoIP channels, combined with slow responding FAX boxes, can lead to a FAX machine hearing its own output heavily delayed, and it may mistake this for the response from the far end. T.38 largely avoids this kind of issue. The start of a FAX call doesn't really have a good signal on which to train a canceller. They can use the first V.21 burst in each direction (The FAX signals for G3 or the V.8 exchange for Super G3), and then lock down the canceller, but those signals aren't wide band enough to be ideal. The canceller could adapt very oddly. If they continue adapting once the wideband signals from the fast modems start, they are likely to upset modem operation there. If they just accept that, and rely on the fast modem retrying, it will usually step down in speed. I believe I have seen this behaviour in setups where the signal looks very clean, but the FAXes always exchange at 12000bps. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do not disturbe
Flavio Miranda wrote:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
This is mine:
[dnd]
;***
;* Do not disturb can be set via Asterisk
;* instead of the phones by dialing this
;* number.
;***
exten = 79*,1,Set(DND=${DB(DND/${CALLERID(num)})})
exten = 79*,n,GotoIf($[${DND} = YES]?3:100)
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
On Tue, 4 Jan 2011, Andy Graybeal wrote: The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone would be best being used as an intercom in a busy kitchen. I asked this some months ago; but this time around I'm writing it into this years budget. I see the 335 has HD Voice and the 321 has Clarity by Polycom. Which would be best in a noisy kitchen using the devices speaker phone? Should I seek another device for the kitchen all-together? I would. The whole Polycom line seems designed for desktop use, and the speakers just don't get very loud. I have especially had this complaint about the ring volume, even at some desktops! In the hotels where we have installations that include busy kitchen extensions there seems to be no substitute for an old analog wall mount phone with a really loud ringer (backed by an ATA). That doesn't help you with intercom though... j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# Hope this helps, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do not disturbe
El 04/01/11 18:13, Flavio Miranda escribió:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
exten = *11,2,GotoIf($[${DND} = YES]?*11,3:*11,101)
exten = *11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)
exten = *11,4,Playback(beep)
exten = *11,5,Hangup()
exten = *11,101,Set(DB(ddisturbe/${CALLERIDNUM})=YES)
exten = *11,102,Playback(beep)
exten = *11,103,Hangup()
I am testing with a softphone and when I dial *11, I receive the
following log from cli:
Executing [...@a2billing:1] Set(SIP/2015-0187, DND=YES) in new
stack
-- Executing [...@a2billing:2] GotoIf(SIP/2015-0187,
1?*11,3:*11,101) in new stack
-- Goto (a2billing,*11,3)
1?*11,3:*11,101
The first 1 before the question mark tells you that the conditional
was evaluated true.
-- Goto (a2billing,*11,3)
This tells you that the goto was been done because the true condition on
the GotoIf instruction.
Everything else goes just as written.
Maybe you need to check some dialplan logic?
Hope it helps.
-- Executing [...@a2billing:3] Set(SIP/2015-0187,
DB(ddisturbe/)=NO) in new stack
-- Executing [...@a2billing:4] Playback(SIP/2015-0187,
beep) in new stack
-- SIP/2015-0187 Playing 'beep.gsm' (language 'en')
-- Executing [...@a2billing:5] Hangup(SIP/2015-0187, ) in
new stack
== Spawn extension (a2billing, *11, 5) exited non-zero on
'SIP/2015-0187'
Therefore, the facilite is not working!!
What I am doing wrong, could somebody point me out please?!!
Thanks in advanced!!
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
--
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
On Tuesday 04 January 2011 16:15:54 Andy Graybeal wrote: The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone would be best being used as an intercom in a busy kitchen. I asked this some months ago; but this time around I'm writing it into this years budget. I see the 335 has HD Voice and the 321 has Clarity by Polycom. Which would be best in a noisy kitchen using the devices speaker phone? Should I seek another device for the kitchen all-together? I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
IMHO G.722 beats Clarity By Polycom every time. I had an IP335 for review before they launched. The audio quality is the same as the better models (IP450/550/650) only the user interface is different. Very good speakerphone, too. Review here: http://www.mgraves.org/2010/01/review-polycom-soundpoint-ip335-entry-level-hdvoice-ip-phone/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen) From: Andy Graybeal andy.grayb...@casanueva.com Date: Tue, January 04, 2011 4:15 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone would be best being used as an intercom in a busy kitchen. I asked this some months ago; but this time around I'm writing it into this years budget. I see the 335 has HD Voice and the 321 has Clarity by Polycom. Which would be best in a noisy kitchen using the devices speaker phone? Should I seek another device for the kitchen all-together? -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote: On 01/04/2011 12:31 PM, Earl Terwilliger wrote: Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory [snip] Have you installed mpg123 or some other program to handle the mp3 files? I am fairly certain that Asterisk cannot handle mp3 natively (most likely for licensing reasons). Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Tom that was my next step if i could not find a way to just play the wav files. I re-booted the server and that seems to have fixed whatever problem was causing this. stopping and starting asterisk had no affect. earl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
On Tuesday, January 04, 2011 04:29:49 pm bakko wrote: Hi CLI module unload res_musiconhold.so CLI module load res_musiconhold.so or service asterisk restart Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi I tried re-starting asterisk (stoping and starting) many times. No success. I finally decided to re-boot the server and that seems to have fixed it. thanks earl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do not disturbe
I really would like to understand why dont works!
should I to set up any other function? maybe on features?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Tue, 4 Jan 2011 20:08:39 -0500
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do not disturbe
Flavio Miranda wrote:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
This is mine:
[dnd]
;***
;* Do not disturb can be set via Asterisk
;* instead of the phones by dialing this
;* number.
;***
exten = 79*,1,Set(DND=${DB(DND/${CALLERID(num)})})
exten = 79*,n,GotoIf($[${DND} = YES]?3:100)
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
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Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt ]# modinfo crc_ccitt ]# uname -a ]# cat /proc/kallsyms | grep crc_ccitt ]# modinfo dahdi -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH problems (asterisk 1.4.38)
On Jan 4, 2011, at 8:49 PM, Earl Terwilliger wrote: On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote: On 01/04/2011 12:31 PM, Earl Terwilliger wrote: Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory [snip] Have you installed mpg123 or some other program to handle the mp3 files? I am fairly certain that Asterisk cannot handle mp3 natively (most likely for licensing reasons). Tom Hi Tom that was my next step if i could not find a way to just play the wav files. I re-booted the server and that seems to have fixed whatever problem was causing this. stopping and starting asterisk had no affect. earl Ah. I got the impression that you were trying to play MP3 files. I would not recommend that you use MP3 instead of WAV, though. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On Jan 4, 2011, at 7:37 PM, Steve Underwood wrote: It is very normal for many people to chat and then start their FAX machines, especially domestic FAX users with a FAX machine attached to their home land line. If you don't care about those your proposal is OK, otherwise. Well, I was suggesting an OPTION to disable the detection after a specified period, so anyone who wishes to do as you describe could just leave the setting to indefinite if they wished. Having said that, I don't think that sort of behavior is particularly common anymore. More importantly, I would argue that practically all users doing that are, as you mention, using a phone connected to a fax machine, eliminating the usual needs for sending the call to the fax extension. Either way, if defined as an option, users could choose how they wanted it to work. There is no excuse for false detection of FAX tone. It takes a very poor detector to mistake voice for FAX, unless the person is specifically trying to whistle the right tones (which some people are quite good at). While that might be true, false detections are anything but unheard of, so being able to disable detection after the first few moments of a call might be useful. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencer marks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: and []# dmesg -c /dev/null; modprobe dahdi; dmesg; lsmod | grep dahdi -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options
[asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Outlook integration
Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT even if contact exists. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
