Re: [Asterisk-Users] Administrative suggestions
Philipp, Good document, my comments are inline with the parts to which they apply. (and yes, this was a top post, otherwise it wouldn't make sense.) Robert > Hi there, > > mostly based upon list postings I compiled a couple of administrative > suggestions on the Wiki page below. I'd be glad to have this reviewed and > commented: > > http://www.voip-info.org/tiki-index.php?page=Asterisk+administration > > Cheers, Philipp > > > Adminstrative suggestions > > Use a GUI client that's based upon the manger API (like gastman or astman > etc) to obtain an overview of what is currently going on in your PBX. Of > course you should also regularly check the log files in /var/log/asterisk > and watch their size. > ** ** Might be good to give some idea of what to look for in the logs and ** why to watch the file sizes. ** > Separate your PC network from your VoIP network (see also Quality-of- > Service (QoS) issues) > > Remove all uneeded modules from your Asterisk server. For example if you > are only doing ZAP and SIP then specify noload= for MGCP, Skinny in > modules.conf. That reduces risks of potential exploits sleeping in those > modules > > Disallow users to work on your Asterisk server. The recently published > serious kernel exploits all required local user access to start with. > > Consider to not use mpg123 for music-on-hold (MOH), or take provisions to > kill hung mpg123 threads whenever applicable. mpg123 has the habit to not > terminate after stopping Asterisk. > ** ** Might want to list the alternatives. Its not a good idea to say don't ** use something without giving an alternative. Personally I don't remember ** seeing that mpg123 shouldn't be considered for MOH. The problem with ** not stopping the threads after Asterisk is terminated might be right but ** doesn't seem to justify a reccomendation not to use mpg123. ** ** > Look into your startup script and take provisions to detect and restart > and hung asterisk. Check out daemontools for this purpose. You could also > regularly telnet into Asterisk (manager.conf) to at least make sure it > hasn't completely crashed. > ** ** "restart and hung asterisk.". Did you mean "restart a hung asterisk process"? ** > Find out if you can run Asterisk with a user other than "root". The > documentation states that in principle that should be possible, however > there seem to be no/few users who have ever attempted this. > > Think about creating your extensions.conf, sip.conf and voicemail.conf > based upon a database that can be shared like mySQL (or whatever else you > are used to). The recently added ODBC support in Asterisk opens up a lot > of possibilites. Next to that the #include syntax that permits to include > other files into any of the .conf files can be of help. > > An unthoughtful change to extension.conf can have a disastrous effect on > your entire PBX. Establish a procedure for those changes to be not > suddenly left without e.g. emergency services (911 or 999 or 112) without > you noticing. Always check the log file after having applied a change to > extensions.conf in a production system. > > Think about putting a quota on voicemailboxes, or schedule a script that > deletes all voicemail older than x days. One way to enable quotas is to > trigger an AGI script just before a user is directed to voicemail and > then decide if a message can be recorded or of the user has run out of > space. > > Use Ethereal (with the IAX plugin) to analyse your network traffic. > > Set an AbsoluteTimeout value for all cost-producing calls to prevent sky- > high bills in case something should ever go wrong with either Asterisk or > one of your phones. Take especially the SIP protocol and its limitations > to detect a disconnected client into account. > ** ** Do you really want to do this and then have people complain because ** a call to an important customer was disconnected? Some conference calls ** can go on for hours, particularly between Germany and the USA due to ** reductions in travel authorizations. ** > Regularly restart (better: stop and start) your PBX during off-hours. A > repetitive reload will not be sufficient, and can actually cause more > harm than it does good. > ** ** Might want to explain why a repetitive reload can "cause more harm than it does good." ** > Spend some thought on redundance, load balancing and maybe even > clustering. So far there is not perfect solution worked out for Asterisk, > however that should not prevent you from thinking about this issue (a > search on the mailing list asterisk-users will reveal a lot of competent > postings) > > * * * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Administrative suggestions
Philipp, Good document, my comments are inline with the parts to which they apply. (and yes, this was a top post, otherwise it wouldn't make sense.) Robert > Hi there, > > mostly based upon list postings I compiled a couple of administrative > suggestions on the Wiki page below. I'd be glad to have this reviewed and > commented: > > http://www.voip-info.org/tiki-index.php?page=Asterisk+administration > > Cheers, Philipp > > > Adminstrative suggestions > > Use a GUI client that's based upon the manger API (like gastman or astman > etc) to obtain an overview of what is currently going on in your PBX. Of > course you should also regularly check the log files in /var/log/asterisk > and watch their size. > ** ** Might be good to give some idea of what to look for in the logs and ** why to watch the file sizes. ** > Separate your PC network from your VoIP network (see also Quality-of- > Service (QoS) issues) > > Remove all uneeded modules from your Asterisk server. For example if you > are only doing ZAP and SIP then specify noload= for MGCP, Skinny in > modules.conf. That reduces risks of potential exploits sleeping in those > modules > > Disallow users to work on your Asterisk server. The recently published > serious kernel exploits all required local user access to start with. > > Consider to not use mpg123 for music-on-hold (MOH), or take provisions to > kill hung mpg123 threads whenever applicable. mpg123 has the habit to not > terminate after stopping Asterisk. > ** ** Might want to list the alternatives. Its not a good idea to say don't ** use something without giving an alternative. Personally I don't remember ** seeing that mpg123 shouldn't be considered for MOH. The problem with ** not stopping the threads after Asterisk is terminated might be right but ** doesn't seem to justify a reccomendation not to use mpg123. ** ** > Look into your startup script and take provisions to detect and restart > and hung asterisk. Check out daemontools for this purpose. You could also > regularly telnet into Asterisk (manager.conf) to at least make sure it > hasn't completely crashed. > ** ** "restart and hung asterisk.". Did you mean "restart a hung asterisk process"? ** > Find out if you can run Asterisk with a user other than "root". The > documentation states that in principle that should be possible, however > there seem to be no/few users who have ever attempted this. > > Think about creating your extensions.conf, sip.conf and voicemail.conf > based upon a database that can be shared like mySQL (or whatever else you > are used to). The recently added ODBC support in Asterisk opens up a lot > of possibilites. Next to that the #include syntax that permits to include > other files into any of the .conf files can be of help. > > An unthoughtful change to extension.conf can have a disastrous effect on > your entire PBX. Establish a procedure for those changes to be not > suddenly left without e.g. emergency services (911 or 999 or 112) without > you noticing. Always check the log file after having applied a change to > extensions.conf in a production system. > > Think about putting a quota on voicemailboxes, or schedule a script that > deletes all voicemail older than x days. One way to enable quotas is to > trigger an AGI script just before a user is directed to voicemail and > then decide if a message can be recorded or of the user has run out of > space. > > Use Ethereal (with the IAX plugin) to analyse your network traffic. > > Set an AbsoluteTimeout value for all cost-producing calls to prevent sky- > high bills in case something should ever go wrong with either Asterisk or > one of your phones. Take especially the SIP protocol and its limitations > to detect a disconnected client into account. > ** ** Do you really want to do this and then have people complain because ** a call to an important customer was disconnected? Some conference calls ** can go on for hours, particularly between Germany and the USA due to ** reductions in travel authorizations. ** > Regularly restart (better: stop and start) your PBX during off-hours. A > repetitive reload will not be sufficient, and can actually cause more > harm than it does good. > ** ** Might want to explain why a repetitive reload can "cause more harm than it does good." ** > Spend some thought on redundance, load balancing and maybe even > clustering. So far there is not perfect solution worked out for Asterisk, > however that should not prevent you from thinking about this issue (a > search on the mailing list asterisk-users will reveal a lot of competent > postings) > > * * * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Development Process comment and Email list suggestion
It looks like Mark and others have addressed the development/CVS issues. We should let their plan be put into effect and give it a chance to work. Regarding the email list: A number of people have suggested creating more email lists. I think this is not a good idea because there will be even more cross posting than there is now between -dev and -users. Several years ago I was a moderator on the SAP-R3-L list hosted by MIT. This list had several thousand subscribers and averaged 300 messages or so per day.This list was run on the ListServe software from Lsoft. The various modules of R/3 (ie: topic areas) were managed by using (and inforcing) topic keywords in the subject line. Enforcing the list guidelines was a 2 hour or so task each night for the moderators (we had several that rotated weekly shifts) but it made the list usage fairly easy. Subscribers could pick the topics of interest and then (assuming people followed the guidelines) their messages from the list were restricted to those topics. Dealing with chronic violators was always a judgement call by the moderator-on-duty: if a person was INTENTIONALLY not following guidelines then they were blocked. If they didn't understand the guideline then it was explained again to them and life went on. oK:: lots of bla bla... My suggestion for the email list is that Asterisk-Users adopt something like the above. Going this way then it would be a much easier experience to join one list and email multiple topic areas (end subscriber gets only 1 email) than several lists where 30% of the messages are cross-posted and subscribers get duplicate email and have to remember which list they need to use for replying. There would definately be a cost for the ListServe license (since there is a commercial profit from the list) but I think this software is best able to handle what we need to do. If Mark and Digium want to go in this direction I would be glad to coordinate the moderation. Regards, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USA dial plan
> Hi, > > Do the callers in USA dialling from USA Telco lines always have to > prefix the CITY/AREA code with "1" in order > To successfully make a call to other USA destinations? > > > I have not been to USA (yet) :) > > Ta > SJ For comprehensive info by area code (and as pointed out it does differ from location to location) check the North American Numbering Plan website at http://www.nanpa.com/. Left menu click on Dialing Plan and then go to the location of your choice. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forums Need Help
> Morning All, > > I have created some virgin forums that I think may relinquish the mailing > lists from major burdens. Everything is .001 in version and I need help. > > I need some advice as far as images and content. I know the project is > opensource but is content and graphics? If not can you help? I am no > artist and need a main page and a good logo. Can I use FAQs that other > people have created as long as I cite them and publish everything in > entirety? > > check it out at www.asteriskhelpdesk.com/forums > > DNS works here but may have not fully propagated to the world. > > Thanks, > Steve Totaro Steve, I am sure your efforts are appreciated but you need to know that there have been at least 2 Forums started but got very little activity... Most of the postings were from the forum owner. I would suggest that you talk with Digium about lending your energy to helping with Asterisk.org or the documentation project. Regards, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] crontab
Philipp von Klitzing said: > oHi! > >> Ladies and Gentlemen, can anyone please help and let me know what is >> the way to start Asterisk automatically using a cronjob, thanks > > http://www.voip-info.org/wiki-Asterisk+administration > > Philipp > > Guess maybe I don't leave my system running long enough for it to crash but seems to me that if the Asterisk process is crashing that we should fix the reason it stops and not just keep on restarting it. On the WiKI there are some writeups of fairly large installations. Are they also crashing? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */
admin said: > I work for an interconnect that sells 3com and NEC. When I made this > project my own and followed through to show my boss, he said, "this is > going > to ruin our industry" > > If that is the case then so be it. Same with mp3s and the music industry. > Had they embraced the technology, everyone could be making a living. Now > they have to sue as a last fight on the way out. > > Really, this is like a car that doesnt run on gas. Seems like it isn't going to ruin your industry but will put a dent in 3Com and NEC !! In fact it could improve your company's business model since you sell services to setup and configure Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chagres Technologies, Inc
John, Take your discussion off list... It is way off topic. I think you do yourself more harm than good by responding to these issues on list. If you want to build confidence in your company then ask your satisfied customers to reccommend you and give their testimonials regarding your speedy service and support. BUT don't get into these arguments. Respectifully Robert John Brown (CV) said: > rick, i didn't say that they couldn't have sent email > from another location. certainly yahoo and msn are > harder to deal with. > > Yes, rick you can do some tracing the way I mentioned. > > lets see:dig routers.com mx > routers.com.4H IN MX10 texas.routers.com. > > ;; AUTHORITY SECTION: > routers.com.4H IN NSdns.inetnebr.com. > routers.com.4H IN NStexas.routers.com. > > ;; ADDITIONAL SECTION: > texas.routers.com. 2D IN A 206.222.193.73 > dns.inetnebr.com. 2D IN A 199.184.119.1 > > > hmm, so i would expect to see email from texas.routers.com > or from some device within 206.222.193.xxx at a min. > (which would cover your machine called vegas.) > > I would expect that a grep -i routers.com mailbox > would produce output that showed mail from that domain. > > I would expect that a grep 192.222.206 db.rbl.ct > would either show something in that block as being a > locatlly flagged spam source, or show nothing, which means > we didn't block it > > i would expect that a grep 206.222.192 /var/log/security > (freebsd ipfw logs) would show something since we have a > rule called permit log tcp from any to mailserver 25 > > I would expect that a grep routers.com /var/log/maillog > which logs the smtp sessions to show something. > > If I didnt' get a hit on any of those, I think its pretty > safe to say I didn't get the email, something is broken. > > All I'm saying is that based on the information we have > for that customer, I can and do check our logs to see if > something got dropped. > > > Almost half of the customers that had issues have their > IP's listed in multiple different RBL's So instead of > dropping those emails, now we have to put them into > a seperate folder and manually check them. we get close > to 2800 spam messages a day into those folders. > > If they sent from hotmail or yahoo, then about the only > thing I can do is grep for there email addy string. > > and yes rich, i'm involved with security issues as > well and have a clear understanding of how packets > move, and what tools I have on my network that allow > me to see whats happening. > > cheers > > > On Sat, Jan 10, 2004 at 03:37:15PM -0600, Rich Adamson wrote: >> common on john, stop the bs. we all know email can be sent >> from hundreds of different valid accounts that you can't >> trace that way (yahoo and msn as just two), and those of >> us that have been involved with security understand it >> rather well. >> >> >> > its simple, >> > i can lookup the MX for his zone, then look up the >> > A RR for each MX, and then search the logs for IP's >> > or I can even expand the search to look for CIDR prefixes. >> > >> > I can also lookup in my private RBL, any query my SMTP >> > machine would have made to see if his IP(s) are spam >> > sources or not. >> > >> > If I don't see packets from those IP(s), or from >> > his MX's, or from his domain, then I'm going to assume >> > no packets where received. >> > >> > cheers >> > >> > >> > On Sat, Jan 10, 2004 at 03:23:13PM -0500, admin wrote: >> > > Sorry, but how can you ID his inbound packets? >> > > >> > > >> > > - Original Message - >> > > From: "admin" <[EMAIL PROTECTED]> >> > > To: <[EMAIL PROTECTED]> >> > > Sent: Saturday, January 10, 2004 3:17 PM >> > > Subject: Re: [Asterisk-Users] Chagres Technologies, Inc >> > > >> > > >> > > > Just refund the guy his money... >> > > > - Original Message - >> > > > From: " John Brown (CV)" <[EMAIL PROTECTED]> >> > > > To: <[EMAIL PROTECTED]> >> > > > Sent: Saturday, January 10, 2004 2:46 PM >> > > > Subject: Re: [Asterisk-Users] Chagres Technologies, Inc >> > > > >> > > > >> > > > > For the list, >> > > > > >> > > > > Mike received a partial order shipped 15-Dec, SN ending 4CD8. >> > > > > >> > > > > Mike received email replies on 3-Dec and 17-Dec advising him >> > > > > on his order. >> > > > > >> > > > > Mike ack'd those emails. >> > > > > >> > > > > This is the first time we have heard anything (phone calls or >> email) >> > > > > from Mike since 17-Dec. Our CDR and SMTP logs show no inbound >> packets >> > > > > or calls. >> > > > > >> > > > > Mike has been sent a private email and has been advised that >> > > > > we will be issuing him a refund on product not received. >> > > > > >> > > > > I can only say that there is a human that answers the phones >> > > > > at Chagres M-F 9-5 MDT (GMT-7). >> > > > > >> > > > > I think I'll change the Auto-Attendent so that it says >> > > > > "For a Human press 0",
Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)
Chandra said: > i also had the same problem temporarily i solved my problem with both > outside NAT. u can also do it if both inside NAT. * outside NAT and > Budgetone behind NAT simply doesn't seem to work. if u ever solve this > problem please let me know too. > > thanks > > cm > I am able to use my Grandstream very well from behind NAT going to FWD. It seems that a proxy server is needed outside NAT in order for SIP to work correctly. Havn't heard of any SIP phones that can jump through NAT without a problem IAX, on the other hand, seems to work fine. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP->PSTN service recomendation?
Chris Albertson said: > > I'm looking for a service that will accept VOIP calls and > send them to the PSTN. Or, I should say _another_ service > that will do this. I don't need the other direction > > Currently I'm using IconnectHere and it works, but I get > complaints of poor audio quality from the other end. But > it sounds OK on my side. > > I like Iconnect's price model: very low/no monthly fee with > a samll per minute charge. I'm in US 310 area code and call > to US, Japan and Canada. > > Who are Iconnect's competitors? Anyone want to recommend > a service? > > Chris, I have been testing http://www.telappliant.com/ They are in the UK but you might find that their rates to the USA are close to some of the USA poviders. I've gotten quality on the calls that meets or in some cases exceeds pure PSTN quality. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New sounds also now in CVS
John Todd said: > >... > Ideas welcome for more text; I may have another timeslot with Allison > early next week in which there will be some leftover room for > additional words. Short phrases and meaningful sets of words for > existing applications are desired; please don't give me words for > apps that aren't even thought out yet. > > John, Thanks for initiating and coordinating the sound generation. This is a good example of how a community can share and achieve more than a single person. But it does take those single people to be the catalyst. A suggestion for the next list would be "good". This could be paired with the already existing morning/afternoon/evening.gsm files to form a greeting. I did not see that combination in the existing files. Regarding the CVS... Maybe could there be an additional directory for all of the weather related stuff (like digits, letters, silence?). Having speciality phrases kept together may keep the sounds directory from becoming so large. Would also make it easier to find phrases. Just an idea. Regards and thanks again for your contributions. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WANTED: Toll-Free gateways in Europe/Asia/Africa/South America
John Todd said:
>
> The freenum.org project wants to use your trunks! The freenum.org project
> is an ENUM parallel tree, which has as an eventual goal the distribution
> of ENUM numbering in nations or areas which due to political or other
> issues are not able to get secure, inexpensive, or functional ENUM
> capabilities.
>
>
my sip.conf contains:
[general]
search => freenum.org
search => e164.arpa
Extensions.conf
exten => _1800X.,1,EnumLookup(${EXTEN})
exten => _1800X.,52,Goto(long-distance,${EXTEN},1)
exten => _1800X.,102,Goto(long-distancei2,${EXTEN},1)
exten => _1800X.,2,GotoIf($[$[${ENUM:0:3} = SIP}] |$[${ENUM:0:3} =
IAX]]?3:4)
exten => _1800X.,3,Dial(${ENUM})
exten => _1800X..,4,Goto(normal-dial,${EXTEN},1)
exten => _1800X.,104,Goto(normal-dial,${EXTEN},1)
I setup my sip.conf and extensions.conf as above. EnumLookup seems to get
a zero return code but the priority that gets executed is 102.
Messages in the console.log are:
Executing EnumLookup("[EMAIL PROTECTED]/3", "18005551212") in new stack
ENUM got '0'
-- Executing Goto("[EMAIL PROTECTED]/3", "long-distance2|18005551212|1") in new
stack
-- Goto (long-distance2,18005551212,1)
Any ideas on how to debug this problem??
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WANTED: Toll-Free gateways inEurope/Asia/Africa/South America
Top posting(sorry) then imbedding the answers to your questions. Otherwise
doesn't make sense.
Thanks for your reply. Sorry it took a while to get the answers. I'm in
Germany and your email came last night just as I was headed to the rack.
Robert
John Todd said:
>>
>>>
>>my sip.conf contains:
>>
>>[general]
>>search => freenum.org
>>search => e164.arpa
>>
>>
>> Extensions.conf
>> exten => _1800X.,1,EnumLookup(${EXTEN})
>> exten => _1800X.,52,Goto(long-distance,${EXTEN},1)
>> exten => _1800X.,102,Goto(long-distance2,${EXTEN},1)
>> exten => _1800X.,2,GotoIf($[$[${ENUM:0:3} = SIP}] |$[${ENUM:0:3} =
>>IAX]]?3:4)
>>
>> exten => _1800X.,3,Dial(${ENUM})
>> exten => _1800X.,4,Goto(normal-dial,${EXTEN},1)
>> exten => _1800X.,104,Goto(normal-dial,${EXTEN},1)
>>
>>
>>I setup my sip.conf and extensions.conf as above. EnumLookup seems to
>> get
>>a zero return code but the priority that gets executed is 102.
>>
>>Messages in the console.log are:
>>Executing EnumLookup("[EMAIL PROTECTED]/3", "18005551212") in new stack
>>ENUM got '0'
>> -- Executing Goto("[EMAIL PROTECTED]/3", "long-distance2|18005551212|1") in
>> new
>>stack
>> -- Goto (long-distance2,18005551212,1)
>>
>>
>>Any ideas on how to debug this problem??
>>Robert
>
>
> Add a priority 2 that looks like:
>
> exten => _1800X.,2,NoOp(${ENUM})
>
> This will allow you to see what gets handed back from the lookup.
> Remember to correctly increment the priorities below 2 to reflect the
> new line.
>
**
** As prty 2 it didn't get executed so made it Prty 102 and got:
** -- Executing EnumLookup("[EMAIL PROTECTED]/5", "18005551212") in new stack
** -- Executing NoOp("[EMAIL PROTECTED]/5", "") in new stack
**
>
> 1) Your priority 2 is missing a space after the | expression. It
> should look like:
> exten => _1800X.,2,GotoIf($[$[${ENUM:0:3} = SIP}] | $[${ENUM:0:3} =
> IAX]]?3:4)
>
**
** space has been added
**
> 2) Priority 102 says "long-distancei2" - is that "i" supposed to be in
> there?
**
** Its not in the actual file.
**
>
> 3) Type "dig NAPTR 2.1.2.1.5.5.5.0.0.8.1.freenum.org." on a command
> line and let's see the answer. You should see the NAPTR replies.
>
**
**
;; QUESTION SECTION:
;2.1.2.1.5.5.5.0.0.8.1.freenum.org. IN NAPTR
;; ANSWER SECTION:
2.1.2.1.5.5.5.0.0.8.1.freenum.org. 21600 IN NAPTR 100 10 "u"
"E2U+voice:sip" "!^\\+1800(.*)$!sip:[EMAIL PROTECTED]" .
2.1.2.1.5.5.5.0.0.8.1.freenum.org. 21600 IN NAPTR 100 10 "u" "E2U+sip"
"!^\\+1800(.*)$!sip:[EMAIL PROTECTED]" .
;; Query time: 877 msec
;; SERVER: 192.168.0.1#53(192.168.0.1)
;; WHEN: Mon Jan 19 20:33:48 2004
;; MSG SIZE rcvd: 195
*
*
> 4) Fire up tcpdump during a test like this: "tcpdump -n port 53"
> and watch as the queries go out and come back.
>
20:30:02.633174 192.168.0.110.1025 > 192.168.0.1.domain: 8485+ NAPTR?
2.1.2.1.5
.5.5.0.0.8.1.e164.arpa. (49) (DF)
20:30:06.837144 192.168.0.1.domain > 192.168.0.110.1025: 8485 NXDomain
0/1/0 (1
10) (DF)
20:30:06.841347 192.168.0.110.1025 > 192.168.0.1.domain: 8486+[|domain] (DF)
20:30:07.016277 192.168.0.1.domain > 192.168.0.110.1025: 8486
NXDomain[|domain]
(DF)
Hmm.. This is different than the DIG. 192.168.0.1 is my Netgear router.
Maybe it is responding to the DNS query from the software and not passing
it the same things that get passed back to DIG. /etc/resolv.conf has
"nameserver 192.168.0.1" so would think that DIG and the software is using
the same source.
> 5) Your priority 4 has two "." characters in the extension regexp.
> That will cause miscellaneous failures.
>
**
** Was ok on the system. In the original email I had to retype so
** introduced typos. Figured out how to cut/paste that stuff now
** from PuTTY.
***
>
> JT
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WANTED: Toll-Free gateways inEurope/Asia/Africa/South America
Looks like the list server is really lagging tonight. I found out some more info so will just post it in a new email with the same subject. I added: "search => freenum.org" to enum.conf and got a match (SIP system) when doing the lookup Maybe I overlooked that in the original instructions. Now will work on trying to get only IAX responses since SIP is rather problematic from behind the NAT router. IAX should work fine. John, Thanks for the tips on debugging. It pointed me in the right direction. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WANTED: Toll-Free gateways inEurope/Asia/Africa/South America
John Todd said: > > Robert - >IAX as a protocol is completely dependent on the far-end gateway, > and not on any specifications you can change. All the gateways at > the moment only support SIP; none support IAX or IAX2, though > hopefully that will change since some of them are actually running > Asterisk as the media gateway. > >As soon as they offer IAX in addition to SIP, then we'll also need > to re-examine the way that Asterisk handles ENUM lookups since > currently only one NAPTR is handed back to the dialplan. For those > nations that have multiple gateways or providers, I have put all the > entries in a round-robin fashion so that the answers will be rotated > by most standard DNS resolver libraries. However, this quickly > becomes unworkable with multiple responses with different protocols, > and there is already a "preference" factor built into NAPTR records > that should be accessible from the dialplan when an EnumLookup is > returned. > >Anyone want to take a swing at it? Otmar? :-) > > JT > John, Thanks for the info. I'll leave the code commented out in the dialplan. If I put in the NAT SIP patch then will reenable it. Is an interesting concept for some long snowy night reading! BTW... regarding the sound directories. I didn't come up with any solutions that haven't already been discussed here. Thanks again for your work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toll-Free Gateway Beta Test: freenum.org
John Todd said: > > > United States:* +1-800-... > +1-888-... > +1-877-... > +1-866-... > via: Telesthetic/Local Exchange Carriers of Michigan > > JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has 8xx dialing into the USA would it be possible to have that one in freenum.org as the returned gateway fro the USA toll free? That way, IAX will get some testing too. Just a thought. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Couple of Newbie Questions: Scrolling, SIP registration, etc.
Info based on how I do it is imbedded below. Robert Larry Keyes said: > I've got two Grandstream phones talking to * and a X100P card going, so > that > I can make inbound and outbound calls via the PSTN, and calls from one > extension to another. > > 1. Is there an equivalent to the "more" command at the CLI to scroll only > one screen at time. How about a way to page the messages scrolling past > when > asterisk is running? > > 2. sip show peers shows my Grandstream phones, but sip show resgistry > doesn't show anything...shouldn't the local SIP phones be registering > with > *? > ** ** As far as I know "sip show registry" shows connections you have made to other servers via SIP (example: gateways) using the register= command in sip.conf *** > 3. I ran the script for creating mailboxes, but don't see these in the > voicemail.conf in /etc/asterisk. ** ** That only creates the mailbox directories and default files in /var/spool/asterisk/sounds/voicemail ** You have to edit voicemail.conf yourself. ** > > Thanks. > > -- L > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK BT Interface with asterisk?
Kannaiyan Natesan said: > Have anyone tried to interface BT's Broadband Voice with asterisk? > > Kannaiyan > ___ > No, and not sure of their rates but http://www.telappliant.com/ has good rates, voice quality and is easy to interface to Asterisk. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK BT Interface with asterisk?
Kannaiyan Natesan said: > Do they offers, free evening and weekend calls? I get from BT. > You can get a free 0870 number from http://www.speak2world.com but they > charge for it. > > Kannaiyan > Don't think so but sometimes "free" isn't free. Depending on calling patterns it might actually be lower cost to pay by the minute with a low per minute rate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)
John Todd said: > > Time to dump the Netgear router. That's an unacceptable answer for a > router vendor to say "Oh, well, for this MAJOR protocol we're going > to simply corrupt those packets so they're unusable." What!? > > JT > __ OR get an older one from eBay. Sounds like in Netgear broke something in the newer routers. I have an RT311 (around 5 years old) that has been working great (knock on wood). My SIP phones work fine with FWD using an outbound proxy. Havn't applied the SIP patch to Asterisk yet but from what I've read that will solve the Asterisk SIP over the router problem. Need to do that anyhow in order to do the ENUM lookup tests. Had thought about getting a new "Modern" Netgear router but guess I'll just keep my trusty old one!!! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some SIP Setup problems
Mike Nash said: > Hi > > I'm trying to configure my Asterisk box to provide a simple sample > configuration. It's a mandrake 9.1 box, no cards except a sound card. > The > config I am trying to achieve is simply one server, with two SIP clients. > > Two issues are cropping up - the first, when I start Asterisk, the sound > goes > nuts and I get an error (below) > > Jan 25 18:16:44 WARNING[163851]: chan_oss.c:268 sound_thread: Read error > on > sound device: Resource temporarily unavailable > > When Asterisk starts, I get the error (below): > > Jan 25 18:06:33 WARNING[81926]: chan_sip.c:446 __sip_xmit: sip_xmit of > 0x80db77c (len 459) to 0.0.0.0 returned -1: Invalid argument > > I'm pretty confident this second error is because I have misconfigured > extensions.conf and sip.conf, but I can't see why. When I try to connect > to > the server with an XTEN client, I get this error: > > Jan 25 18:15:38 NOTICE[81926]: chan_sip.c:5548 handle_request: > Registration > from 'Mike ' failed for '203.118.186.16' > > I've tried looking at the www.automated.it setup information, along with > the > information on fnords.org - this has gotten me this far, but I can't see > for > the life of me what I have done wrong. > > If anyone could provide me some pointers, it would be much appreciated. > > Regards > > > Mike > > My SIP conf looks like this: > > ; > ; SIP Configuration for Asterisk > ; > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0; Address to bind to > context = sip ; Default for incoming calls > > [Phone1] > type=friend > secret=yap > auth=md5 > nat=yes > host=dynamic > dtfmmode=inband > mailbox=1000 > username=mike > context=sip > disallow=all > allow=gsm > callerid="Mike Nash" <6969> > > [Phone2] > type=friend > secret=yap > auth=md5 > nat=yes > host=dynamic > dtfmmode=inband > mailbox=1000 > username=darryl > context=sip > disallow=all > allow=gsm > callerid="Darryl West" <6970> > > > > My extensions.conf looks like this: > > [general] > ; > ; If static is set to no, or omitted, then the pbx_config will rewrite > ; this file when extensions are modified. Remember that all comments > ; made in the file will be lost when that happens. > ; > ; XXX Not yet implemented XXX > ; > static=yes > ; > ; if static=yes and writeprotect=no, you can save dialplan by > ; CLI command 'save dialplan' too > ; > writeprotect=no > ; For more information on applications, just type "show applications" at > your > ; friendly Asterisk CLI prompt. > ; > [sip] > exten => 1,1,Dial(SIP/Phone1,20,tr) > exten => 2,1,Dial(SIP/Phone2,20,tr) > exten => 1000,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) > Mike, Are you using the 0.7.1 tar distribution or CVS? I was able to compile the 0.7.1 Asterisk program/sample config's to get a working system on a PC with no sound device and no phone interfaces. This system is about as simple as it can get (except for the 3 fixed disks in it) and is even a low end Pentium I (100 Mhz, 32MB RAM). My suggestions are: - delete the source directory (including /etc/asterisk) and rebuild Asterisk (make install) and the sample config. (instructions on screen after the Asterisk build). - try to start Asterisk with NO changes to the config files. If it doesn't start then the problem is probably with your system hardware. - copy one of the device configs in sip.conf and change for one of your phones. You should only have to change the device id. Don't get fancy with authentication or NAT. (assuming that Asterisk and your first phone are not on OPPOSITE sides of a NAT Router) - either issue the RELOAD command at the command line interface or completly shutdown and restart Asterisk. (RELOAD should be sufficient but one of the 2 is required) - test that phone by calling extension 1000 (assuming you have the 0.7.1 extensions.conf). You should get the demo greeting and be able to do such things as ECHO test, leave voicemail for a sample mailbox. Instructions are in the demo greeting you will hear. - If all of the above works then add your 2nd phone into sip.conf and add a dialplan for those 2 devices to extensions.conf so that they can call each other. (don't forget to RELOAD or restart) - IF all of that works then start making fancy config files and using additional features. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for iax termination
> - Original Message - > From: Daniel Bichara > To: [EMAIL PROTECTED] > Sent: Saturday, January 24, 2004 4:12 PM > Subject: Re: [Asterisk-Users] looking for iax termination > > > Hi, > > We have termination based on IAX and SIP at Brazil. > > Daniel > Daniel, I would be interested in the details of your termination into Brazil. We have several Brazilian expatriates here in Germany that might be interested in your service. Partially would be Asterisk using IAX2 and others using SIP Phones. Can you please pass along additional info? Regards, Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAPRTC load error
I have compiled the zaptel library and zaprtc on a system that gives the following from "uname -a": Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC 2002 i686 unknown Makefile for zaptel had the following line uncommented: # KFLAGS+=-D__SMP__ When doing the "make load" for zaprtc I get the following error: modprobe zaptel /lib/modules/2.4.18-64GB-SMP/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.18-64GB-SMP/misc/zaptel.o was compiled for kernel version 2.4.18-4GB while this kernel is version 2.4.18-64GB-SMP. Any ideas on where to look for the solution would be appreciated. Have checked the Makefiles but didn't see anything related. Thanks, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Words for Allison(?)
Rob Fugina said: > On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: > > > In the mean time, I've seen references to bug #'s, here on the list and > in the CVS logs. I've yet to stumble across the bug tracking system, > though -- can you give me a nudge in the right direction? > > Thanx, > Rob > > http://bugs.digium.com/ Its the first entry in the google result when you search for "asterisk" "bug tracking"!!! You may also want to check out http://www.asterisk.org and the documentation wiki at: http://www.voip-info.org/wiki-Asterisk if you havn't "stumbled " across them yet. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large scale e.g. university
Martin said: > Hello. > > I vaughely remember someone talking about an asterisk implementation at a > University in germany some months back. > > Any other information ? > > Regards...Martin > -- > http://graphics.cs.uni-sb.de/VoIP/en/index.html Some of those folks and also from the Uni Stuttgart hang out on here too. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mark's Asterisk Presentation at Linux-Kongress2003
Real Player is required. Excellent video/slide presentation. http://graphics.cs.uni-sb.de/VCORE/recordings.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Code Hosting...
Andy, I would be interested in your Cepstral engine code. Regards, Robert Friedrichshafen, Germany Andy Powell said: > lo, > > Is there a single central location for code and applications other than > CVS? I'm talking about code that can't/wont be included in CVS for various > reasons? Does the wiki have this sort of thing? I've done some code for > the Cepstral TTS engine (bkw has done some updates too) but apparently > this will never make it into CVS (since the engine is not GPL)... Seems to > make sense to have a central location for this type of 'outlaw' code... > The bug tracker is useless for this sort of thing but there seem to be a > number of bits of code like this in there > > moo > > Andy > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral TTS Code
Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling SIP
Tim Sailer said:
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered SIP to be seen
> as unavailable. Am I just missing something obvious, again?
>
> Tim
> ^
Tim,
I use the following in my dialplan to distinguish between Unavailable (ie:
did not answer), Busy and Channel doesn't exist. ChanisAvail goes to
n+101 if the channel is NOT avail. There is probably a better way to exit
the sequence but that is what works for me.
exten => 11,1,Macro(stdexten,11,SIP/11)
Below is the macro for the above... Have tested it with IAX2, SIP and MGCP.
The first argument is the macro name, 2nd is the voicemailbox, 3rd is the
Channel to dial.
[macro-stdexten]
exten => s,1,ChanisAvail(${ARG2})
exten => s,2,Dial(${ARG2},20,Ttr)
exten => s,102,Voicemail2(u${ARG1})
exten => s,103,Hangup
exten => s,104,Voicemail2(b${ARG1})
exten => s,105,Hangup
LIke I said.. its messy but does work.
Robert
Friedrichshafen, Germany
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Smallest Asterisk Server Ever?
Christian, Where is a good place to purchase your phones in Germany? I found a distributor in the UK but maybe just am not looking in the right place for Germany. Thanks, Robert American Expatriate in Friedrichshafen (Grund oder Entschuldigung für die englisch) Christian Stredicke said: > Sorry, we have to make some money... Product business is tough! > > :-) CS > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid & AGI Thougts
[EMAIL PROTECTED] said: >> > I like using whisper tones... > > recored the file companyname_whisper.gsm and put it in > /var/lib/asterisk/sounds > > Then add the lines to extensions.conf > > exten => 0031,1,Dial(SIP/Recp|20|A(companyname_whisper.gsm)r) > > In my implementation of this the file extension had to be left off. exten => 0031,1,Dial(SIP/Recp|20|A(companyname_whisper)r) Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International PSTN dialing
Matt McIntyre said: > I am interested in subscribing to a service that will let me dial the > PSTN in Ireland and am interested in what the community thinks about who > has the best services available. I would prefer to purchase time in > blocks of minutes or pay as I go in lieu of having a monthly fee to > contend with since I don't plan to need to use it very often. > > Thanks for the help, > > Matt > Matt Check out http://www.telappliant.com/ They are based in the UK. I have a test account with them that has worked really great via IAX. 73 Robert WB4DHC/DL4GRC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Call did not go through"
Jim Sneeringer said:
> Whenever an outside number is dialed, Asterisk says "We're sorry. Your
> call
> did can not be completed as dialed. Please check the number and dial again
> or call your attendant to help you." I have tried many configurations,
> but
> let me give the simplest: It fails when a local number is dialed and the
> context contains only the following rule:
>
> [default]
> exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
>
> where
>
> TRUNK = Zap/1&Zap/2
>
> which are Digium FXO cards.
>
> It works with
>
> exten =>9,1,Dial(${TRUNK}/${EXTEN})
>
> Furthermore, it was working before. To my knowledge, the only thing I
> changed to make it fail was to shut down the working test system, move it
> to
> the actual environment, and make it live. I had been testing with only
> one
> of the two CO lines. Maybe I changed something in extension.conf, but if
> so
> I don't know what it was.
>
> Incoming and intercom calls work fine.
>
> Can anyone tell me what is wrong? Thanks.
>
> Jim
>
In: exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:1}) you strip off the
leading 9
in exten =>9,1,Dial(${TRUNK}/${EXTEN}) you do NOT strip the 9 off.
Could it be that your external line connected to the Digium card is
actually connected to some other system and needs the 9? (and the
message is actually coming from that system) Connect an analog phone to it
and see what dial string it needs.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EMEA and Chagres Technologies
John, You are now advertising your EMEA company in your signature block. Maybe I missed an email that explains the EMEA pricing and availability. Could you please give an update via the list as to the status of your product availablity, pricing and delivery times in Europe? The ordering procedure would be nice too assuming that you are able to deliver to the EU from the EU. Dealing with customs charges for individual shipments from the USA is not desirable. Thank you, Robert John Brown (CV) said: > > . > > > john brown > chagres technologies, inc (Americas) > chagres technologies, b.v. (EMEA) > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP extension "busy" when not available ??
Soren Rathje said: > - Original Message - > From: "Olle E. Johansson" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, February 22, 2004 8:52 PM > Subject: Re: [Asterisk-Users] SIP extension "busy" when not available ?? > > >> > Although the current logic does not require a sip phone to register, >> it > would >> > seem like the asterisk logic should be something like: >> > a. call is attempted to sip x1234, >> > b. if * knows the extn is in use, return busy, or, >> > c. if not busy, asterisk attempts to contact x1234 across the wire, >> > d. if no contact, return Unavailable >> Or if not registred. >> > > In a wireline telephony scenario the above would be the proper method as > we > do not know if the subscriber have their phone plugged in or not. With > Asterisk we experience the same information as mobile operators do: > "unreachable", "unanswered" and "busy". IMHO we should have the same > options. > > So, the priority for a type=friend would be: > a: check if client is registered and/or reachable, if not - return > "unreachable" > b: check if client is busy, if call-waiting not active - return "busy" > c: if call is rejected by client, return approriate message > d: if call is unanswered, return "unavailable" or "busy" with reference to > (b). > > -- Soren I use ChanIsAvail() to check to see if the phone is connected at the top of the dialplan for that extension. This works for IAX2 and SIP channels but not for MGCP. If you are interested in the actual code I can send it to you from home tonight. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP extension "busy" when not available ??
Soren Rathje said:
>>
>> I use ChanIsAvail() to check to see if the phone is connected at the
>> top
>> of the dialplan for that extension. This works for IAX2 and SIP channels
>> but not for MGCP.
>>
>> If you are interested in the actual code I can send it to you from home
>> tonight.
>>
>> Robert
>>
>
> Thank you, yes please...
>
> Well, I'm about three weeks into my very first * installation (that sort
> of
> works), so basically any info/tips/tricks/"word of advice" is accepted
> with
> appreciation...
>
> -- Soren
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension.
The Macro is:
[macro-stdexten]
exten => s,1,ChanisAvail(${ARG2})
exten => s,2,Dial(${ARG2},20,Ttr)
exten => s,102,GoTo(voicemail,s,1)<--Note A
exten => s,103,Hangup
exten => s,104,GoTo(voicemail,s,1)<--Note B
exten => s,105,Hangup
The extensions are defined as:
exten => 10,1,Macro(stdexten,10,MGCP/aaln/[EMAIL PROTECTED])
exten => 11,1,Macro(stdexten,11,SIP/11)
exten => 12,1,Macro(stdexten,12,IAX2/12)
The 2nd argument in the () is the voicemailbox number, 3rd argument is the
Channel to dial.
Note A: If the Channel is not available then control comes here. You can
put a Voicemail2 statement here with the u option or whatever you want to
use.
Note B: This is where the Busy/Timeout comes from the Dial command.
In my case I have a voicemail context that handles the 2 mailboxes we use
here in the house. That is: an announcement is played and the caller
selects the mailbox to get the message.
Its not perfect and for sure can be improved.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need some information
Comments are inline. Robert Jeroen Rikhof said: > Hello, > > Can somebody give me some information about: > > 1. How stable Asterisk is? My experience and from what I have read on the list is that it is very stable if run on stable hardware and you don't mess with the program code. If you mess with the code then all bets are off. (applies to ANY system) > 2. If asterisk crashes, how quick can i get it up? I would think that the answer to this depends on what causes the crash. Probably its hardware related so isn't an Asterisk question but rather one for your hardware provider. > 3. Billing software? (I'm storing CDR in MySQL DB) There are some available. Search the email list and Google. > 4. DO and DO NOT with Asterisk? DO install it and experiment. DO expect that if you want to change functionality then you can change the programs.. (see the above about crashes) DO learn how to search the list archives and Asterisk WIKI. DO utilitze the several websites that have examples. DO NOT ask questions about problems without providing clear documentation of the problem including conf. files and traces where available. DO NOT expect Asterisk to be all things to all people. > 5. Reducing Echo with Budgetone-100? (Now Using Program 1.0.4.40 firmware) My Budgetone doesn't echo... but I am not connecting it to the PSTN. From what I have read, echo is more of a factor of digital to analog. > 6. How can i create roaming users? Depends on what you mean by "Roaming users", from where they connect and how they connect. There are some conf. examples around for people to "log onto" an extension. > 7. What is a good Managing(easy use) program for asterisk? I use the VI program to edit config files. It works great. A good understanding of the config files is a requirement for a successful implementation. Take the time to learn them before you look for some fancy GUI for configuration. > > Kind Regards, > > Jeroen > [EMAIL PROTECTED] > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] record application in extensions.conf -- how to stop recording?
Paul Mahler said: > With record: > > > > ; Record voice file to /tmp directory > > exten => 9000,1,Record(/tmp/asterisk-recording:gsm) > > exten => 9000,2,Hangup > > > > Is there a way to stop recording other than hanging up? > > > > Thanks! Press the # key. Below is from my extensions.conf. It plays the recording back after the # key is pressed. the prty 6 hangs up the call from the Asterisk side. Has nothing to do with what you do on the callING side. (as far as I know). ; Record a temp. GSM file exten => 100,1,Wait(2) exten => 100,2,Record(/tmp/asterisk-recording:gsm) exten => 100,3,Wait(2) exten => 100,4,Playback(/tmp/asterisk-recording) exten => 100,5,Wait(2) exten => 100,6,Hangup Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Angel Gabriel said: > I have 5 BT phone lines coming into my office. We use four for > international calls, and one for local/mobile calls. We have just obtained > another call carrier, and now we would like to be able to make calls from > any phone to any carrier, without having to remember what details to tap > into the phone. I would like all calls to be prefixed with the relevant > codes, so that my employees can all dial direct. Also, incoming calls, I > want them all redirected to just one phone, the one in reception, and then > diverted as required. Is the above possible?? > You will have to have a way of determining which which carrier and/or line should be assigned to each call. Can you explain how you would make that determination? Once that is known then you will know how to build the dialplan to support it. Asterisk can add or subtract digits to/from dialed numbers so its no problem to add the carrier selection codes to the front of the number (assuming it works the same way there as here in Germany and the USA). What do you mean by "diverted as required"? You will need to specify the diversion rules. examples: If operator is busy, ring phones 2,3. If its night then go to voicemail. If the call is from the boss's cell phone then forward it to your cell phone. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple * status
Tim, It looks interesting.. Are you willing to release the source code? Robert Tim Sailer said: > On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote: >> Since there's not too much out there, I decided to take about 2 hrs and >> pound something into shape for a simple status for my * server. >> I wrote a perl script that parsed the output of 'sip show peers', >> 'iax2 show peers', and 'show voicemail users' through the manager >> interface. It dumps the output to a few simple mysql tables, and >> the results are displayed on a web page. Now I can see some of the >> basic things. >> >> http://pbx.unslept.com/status.php > > Before anyone comments, I know it's rough and ugly looking, but this was > just proof of concept for me, done over about 2 hours while trying to do > my normal job, too. I'll keep poking at the CLI to see what other cool > stuff I can pull out. > > Tim > > -- >>< >>> Tim Sailer >< Coastal Internet, Inc. << >>> Network and Systems Operations >< PO Box 726 << >>> http://www.buoy.com >< Moriches, NY 11955 << >>> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >>< > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL and 800 numbers
I have made no recent changes to the IAX2 config on my system. Today I tried a 1800 call and got the below error. Not sure when this started since only use 800 once in a while. Does anyone know if IAXTEL is experiencing problems connecting to the 8xx gateway? 7 16:14:54 WARNING[147466]: chan_iax2.c:4445 socket_read: Call rejected by 69.73.19.178: No such context/extension -- Hungup 'IAX2[69.73.19.178:4569]/3' == No one is available to answer at this time Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile problem SuSE 8.2
I am trying to compile * on SuSE 8.2. When doing the "make install" in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See http://www.gnu.org/software/gcc/bugs.html> for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Results SUSE 8.2 + server size
Hello All, Thanks to those that responded to my problem of compiling on SUSE 8.2. I was not able to get the compile done so decided to put RedHat 9 on this system. After getting a RedHat supported NIC and RedHat installed, Asterisk compiled cleanly, one SIP phone is connected and voice mail works. No other tests have been run yet. A couple of days ago, Michael Farnworth asked about the smallest system that was running Asterisk. This one is a Pentium 100, 32 MB RAM, 8 GB disk. I don't expect it to handle much load but for a test platform it seems ok to use while trying to find a low cost P4 system. Regards, Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ISA tormenta card message]
I am getting the following messages that seem to be coming from Asterisk. In the system there are no ZAPTEL cards installed. I did uncomment ztdummy in the Makefile in /usr/src/zaptel before running "make install". Any ideas on how to get rid of this message. I looked through all the config files (installed the sample ones then modified sip.conf, extensions.conf and voicemail.conf, rest are as installed) but did not find anything that looked right. Can someone please point me toward what I am overlooking? Thanks Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Setup
My config that works for number 1 is below. Everything works including the voice mail waiting light. All of this for * was copied from or based on: http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting started site. Can't help you with #2 but am sure others can. sip.conf for extension 2000 [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it extensions.conf exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) Budge Tone config: SIP Server: 192.168.0.110 (my * box) SIP Userid: 2000 (userid is same as extension Authenticate ID: 2000 Authenticate password: 9overthruster7 Send DTMF: Via SIP info (in order for the dtmf to be recognized by voicemail) > Hi People, > > Ok i've tried everything I can think of but cant get this to work. > > Can someone please give me an example of their sip.conf settings and also > the > details of the settings in their grandstream phone to allow: > 1. Grandstream phone to register with asterisk when on same lan. > 2. Grandstream phone to register with asterisk when phone is behind a nat. > > Regards, > Aaron. > > > > - > This mail sent through IMP: http://horde.org/imp/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proper Credit: Re: Grandstream Setup
I was incorrect in my citation of credit in the below email. Properly the credit goes to John Todd for the Asterisk config examples. His excellent article is at: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1 Sorry for the goof-up. Robert > My config that works for number 1 is below. Everything works including > the voice mail waiting light. All of this for * was copied from or based > on: > http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting > started site. Can't help you with #2 but am sure others can. > > sip.conf for extension 2000 > [2000] > > type=friend ; This device takes and makes calls > username=2000 ; Username on device > secret=9overthruster7 ; Password for device > host=dynamic ; This host is not on the same IP addr every time > context=from-sip ; Inbound calls from this host go here > mailbox=2000 ; Activate the message waiting light if this > ; voicemailbox has messages in it > > > extensions.conf > > exten => 2000,1,Dial(SIP/2000,20) > exten => 2000,2,Voicemail(u2000) > > > Budge Tone config: > > SIP Server: 192.168.0.110 (my * box) > SIP Userid: 2000 (userid is same as extension > Authenticate ID: 2000 > Authenticate password: 9overthruster7 > Send DTMF: Via SIP info (in order for the dtmf to be recognized by > voicemail) > >> Hi People, >> >> Ok i've tried everything I can think of but cant get this to work. >> >> Can someone please give me an example of their sip.conf settings and >> also >> the >> details of the settings in their grandstream phone to allow: >> 1. Grandstream phone to register with asterisk when on same lan. >> 2. Grandstream phone to register with asterisk when phone is behind a >> nat. >> >> Regards, >> Aaron. >> >> >> >> - >> This mail sent through IMP: http://horde.org/imp/ >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
Do you have a 100 or 101? You have indicated different models in your postings. Were you able to get Call Transfer and Call Waiting working with your Asterisk system and other phones? Which version of the Grandstream firmware do you use? There most recent on their website this weekend was at least 2 version numbers higher than what came on my phone in August. Think that they are making improvements rather frequently. Robert > On Wed, 15 Oct 2003, Jon Pounder wrote: > >> >The Grandstream 101 I'm using is a piece of junk but I don't have the >> same >> >problem with it. >> >> What don't you like about the grandstream ? (I am not looking to flame >> you, >> but was considering buying and if there are problems would rather find >> out >> beforehand) > > Nothing works. Call transfer and call waiting, in particular. (Well, > almost nothing; vm notification does work) > > There is no place to plug in a headset, and since I do a fair amount of > tech support and longish conference calls, that's a big deal for me. > > However, keep in mind that I have an old, no-longer-manufacturered model > (the Budgetone 100). Don't take my frustration with my outdated phone as > a sign that you should dismiss Grandstream out of hand - I just don't like > my 100. > > -- > JustThe.net Internet & Multimedia Services > 22674 Motnocab Road * Apple Valley, CA 92307-1950 > Steve Sobol, Proprietor > 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
I only have 1 but the absolutly only time it has to be rebooted is when I change a parameter or upgrade the firmware. It has run for weeks without any problem. Another poster mentioned the 10 vs. 100 Ethernet speed. Maybe Grandstream can upgrade the interface in future hardware. I don't imagine that the price point for 10/100 is much different than 10 these days. One option I would definatly like is the ability to turn off the ringer. Since my testing ususally happens after my wife goes to bed it would help NOT to have the audible ring but only the visual indication! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream ringer
Michael, That would work for me too. If the volume can be reduced (maybe to zero or almost zero) then my request for the ability to disable it is not needed. Since the volume of the speaker and handset can be controlled maybe the GS folks can include a patch in the next release of the firmware to also handle the ringer. They monitor this list so maybe will jump in with some feedback to us. Robert >> Better still I would like volume control over the ringer as the default > tends to be rather loud and annoying to other people in the same room. > > Michael > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Its a free world and everyone is entitled to their opinion. Here's mine on this topic. The cards aren't so expensive (99.95 USD). If they have their own hardware then they don't have to depend on the target system having a particular configuration. Example: right now I am running * on a system that has NO USB ports so couldn't use USB for timing. That makes their programming much easier. As has already been pointed out, nothing keeps someone from writing patches to use some other timing device. Digium is nice enough to put Asterisk as public software. Lets don't screw that up. Be part of the solution, don't complain about the problem. If you have a solution suggestion then post it.. probably others would be happy to help you Robert >> > If the software needs a specialcard to keep time then the >> > software is broken or poorly designed. > >> Don't complain so loudly unless you're willing to contribute the >> fixes. Opinions are like assholes, and you know where that's going. >> Takes something else entirely to fix a perceived problem. > > He is correct though; I am willing to put money on the idea that the > Tiger320 chip is NOT a stratum 1 time device (or even stratum 3 for that > matter) -- Chances are that Digium was trying to tie in their hardware and > make money to support * that way, and you can't really fault them for > that. > > There's quite a difference between an uninformed opinion and the opinion > of > someone who's actually written software and designed hardware that meet > more stringent requirements than what is being discussed, do you not > think? > > I agree though -- there is _no_ reason why the USB, RTC or even regular > PIT > timer can't do this job just as easily... In fact, both USB and RTC > options have been created and seem to work just fine for most people. > Providing hardware to do this also gives an advantage that * can be ported > to multiple platforms with minimal software shuffling, and since drivers > would have to be written for the card anyway, you've got your time source > with you where ever you go. > > There are a number of questions that keep coming up that Digium seems > unwilling to answer. It'd be nice if they just came right out and said > "Guys we need to make money to support * development, this is how we are > proposing to do it..." but instead we get closed lips and speculation. > > It does seem that there are a number of systems out there that have > marginally-compliant PCI busses or some other slightly odd aspect to them > that make jitter and echo on * worse than other systems. It'd be really > nice to get a database of known-bad hardware together. > > Regards, > Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
The only thing that is wrong is that there seems to be some expectation of Digium that they have to tell things... The source code is available. If someone isn't happy with the Digium methods then they should find a solution and post it to the list and/or one of the several Asterisk Wiki's that are around. Digium has no obligation in this regard. OSS doesn't mean free, OSS doesn't mean no secrets. It means Open Source Software. Alot of folks (me included)sometimes incorrectly equate the term OSS to mean FREE. Digium has already given for free much more than the typical telephony hardware manufacturer. I think its pretty clear that they like using "common timing source across all platforms". I am looking forward to the postings of alternative timing solutions. Robert > > Also as I'd written. It seems we're arguing the same side of the > argument. > :-) My compliant was not that the timing was needed, but that Digium > seems so damned secretive about it. I mean this is OSS -- just tell us > that having a common timing source across all platforms makes things > really > easy, you don't have to screw with looking at writing an alternative > driver > for RTC or USB or XYZ and hey, we happen to make some money selling these > boards too. If you are running a SIP-only * box then here are some > alternative timing drivers, point them at some URLs and oh by the way, we > didn't write 'em, we don't support 'em, they seem to work fine with others > though. > > What's wrong with that? > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Andrew, I am running it rather well on a original Pentium 100 Mhz, 32 MB RAM, no USB adapter. I agree with you this would not be an ideal setup for a business but in a home it will work rather well. I think it'll handle 2 CO analog lines fine. Yes, my wife thinks its overkill. Probably is, but guess what, if I want to change it I can. If I want to try and integrate another type of card, I can. If I want to connect and control my ham station, I can. AND best of all, if I want to develop and use another timing source, I can. and so can you. Regards, Robert > > I don't think you'll be running Asterisk on anything older than a P2 to be > honest, and even then the utility is severely hampered due to everything > being done in software on *. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Chris, Good point. As I understand it, the Asterisk software requirement was to be a PBX between normal telephone lines and VoIP. Maybe even it was just to replace the expensive PBXs. As such seems to me that it clearly met and exceeded its design requirements since it utilizes the hardware boards that were in the original design requirement. Don't think anyone can dispute that. Its created by Digium for Digium hardware. Everything lese is "gravy". 73, Robert > > > > Any discusion about PCI cards, RTC timmers and the like is in > a complete vacuum unless you know what exactly it is that the > software is required to do. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question: Meetme
Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As indicated in previous posts I do not have any Digium cards in the system. When making the zaptel part of the system I did uncomment ztdummy.o in the MODULES= line. Extensions.conf contains: exten =>2663,1,Meetme,9876 meetme.conf is: [rooms] conf => 9876 If meetme doesn't work at all without a real card that is ok. It can wait. If it'll work at least somewhat with ztdummy then obviously I've missed something. Any ideas? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme
> On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote: >> Yes, I am a newbie too. I am having a problem with meetme. From what I >> have seen it will work without a Digium card but with audio problems. My >> goal is just to see how it works not the quality of the audio. >> >> When I dial into the conference room the following message is played: >> "That is not a valid conference number." >> >> On the console I get: "unable to open pseudo channel". >> >> As indicated in previous posts I do not have any Digium cards in the >> system. When making the zaptel part of the system I did uncomment >> ztdummy.o in the MODULES= line. > > did you actually install the module into your running kernel? > > -- Steve, I have the following line in /etc/modules.conf: post-install ztdummy /sbin/ztcfg Is that what you mean or did something else need to be done that I missed. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme
> > Did you "modprobe ztdummy" before running asterisk ? I have meetme > running in one * box without zaptel harware. > I just tried that. The following messages are given: /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed ztdummy.o is in /lib/modules/2.4.20-8/misc so I tried: modprobe /lib/modules/2.4.20-8/misc/ztdummy.o and got: modprobe: Can't locate module /lib/modules/2.4.20-8/misc/ztdummy.o Permissions of that file are: -rw-r--r-- I noticed in the Makefile of zaptel that PRIMARY=torisa. Can zaptel be remade (or whatever that is called) without having to redo Asterisk? If so maybe I try PRIMARY=ztdummy. What do you think? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme
The USB card is the problem.. I should have realized that from today's other thread... This system has no USB and unless I find a ISA USB card it won't either since the PCI slots are full. There is plenty else to get familar with so meetme and MOH goes on hold (without music :-) ). Sorry for cluttering up the list on this problem. Thanks for the help. Robert > > These are some pretty simple error messages. It basically says it can't > load the usb-uhci module that ztdummy needs. So either you do not have a > uhci usb device, or you are using the other driver which would then > cause the device to be otherwise busy and not able to answer the calls > for this driver. So you need to know either sort out your usb problems, > or switch to the ztrtc driver. > >> ztdummy.o is in /lib/modules/2.4.20-8/misc so I tried: >> modprobe /lib/modules/2.4.20-8/misc/ztdummy.o > > modprobe consults the modules dependency database, and uses names of > drivers, not filenames. This is why this example failed. > >> and got: >> modprobe: Can't locate module /lib/modules/2.4.20-8/misc/ztdummy.o >> Permissions of that file are: -rw-r--r-- >> >> I noticed in the Makefile of zaptel that PRIMARY=torisa. >> Can zaptel be remade (or whatever that is called) without having to redo >> Asterisk? If so maybe I try PRIMARY=ztdummy. > > As far as I know, PRIMARY is useless since the majority of people here > have probably never even seen that card let alone have it. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I give up!!
> Asterisk... > Linux... > You get what you pay for. And it's free > :P > > Thats true but free (cost) doesn't have to mean cheap (quality). Maybe what we need is to collect business requirements and build a configuration for a typical system. (hardware spec. and actual config files) What Dave has listed is a good start. Then folks will have a starting point. If cost is the driving factor then obviously there has to be a compromise in functionality. Knowing what a specific functionality costs to implement would help people quoting installations. (example the transfer situation that GS phones don't handle but seem to work with one of the more expensive phones and the rest as GS). While I don't have the hardware or even Asterisk knowledge (yet) to do this, I'll be glad to document results in a set of webpages (or maybe we should use one of the already existing sites). Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme (looking for ztrtc)
>> > look at the rtc driver then. you do have a rtc chip already on the > system. > >> I looked back in the list and looks like the message that mentioned who wrote ztrtc I deleted. Can someone please let me know where to obtain ztrtc? I did a google on it and came up empty. Thanks, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme (looking for ztrtc)
>> Seems you used my abreviation. It is really known by zaptelrtc. It seems > to be written by Klaus-Peter Junghanns <[EMAIL PROTECTED]> and is > distributed at http://www.junghanns.net/asterisk/. > Thanks for the info Steve. I got it but the "make" didn't work. Will work on it over the weekend. > > Not trying to stir up old flame wars, and not directed at the person > requesting the information above. > > This was found with a combination of google and grep -ri over my mail > directories. Proof positive that a web only based version of this list > is not a good option. > You are right.. but at least there are archives available. maybe if I had looked for "rtc" and ASterisk then might have gotten a hit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The Start extension
I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the "s" extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten=> s,1,Answer exten=> s,2,Background,dir-intro exten=> s,3,DigitTimeout,3 exten=> s,4,ResponseTimeout,10 exten => 2000,1,Dial(SIP/2000,20) exten => 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Start extension
> > > > The "s" extension is used when there is no known called number. In > other words, if you are dialing 2000, the dialplan will always prefer > the priority list for 2000 instead of going to 's', so that is why > your current system doesn't work. > John, Thanks for the details. Actually what I want to do is to play an announcement and then pass the person along to the extension that they dialed. Use of Background was probably not the correct command. (sb. Playback). YOur details clear up the order of processing. Think I can get it from here. Thanks again. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert > Hi List, > > I had a wonderful meeting with GS's President last week > and he is very interested in feedback on what top features, > functions, bugs the community would like to see in upcoming > firmware. > > Please keep in mind that adding new features take time > to develop, test and such. > > So please rate your ideas on a scale of 1-10 > > 1 = Nice to have some day > > 10 = Got to have it right now > > > > Things like ring tones and fixing call waiting are already > on the list. :) > > Lets also keep the replys away from gripes and complaints > and more towards constructive comments. > > I'll be taking the results and sending GS a summary. > > John Brown, > Chagres Technologies, Inc > > Buy your VoIP hardware from us > email: sales at chagres d0t net for quotes > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
> On Tue, 21 Oct 2003, Low, Adam wrote: > >> Maybe I am missing something here but why would it downgrade their >> network speed to 10mbps, its very rare to find a 100bT switches these >> days that don't also support 10bT. In a switched ethernet network there >> would be no performance loss for the other ports !? > > The cable goes into the phone and then out of the phone into the computer. > That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. > Perhaps the best way to avoid this is to join all the phones together > since they are all 10Mbit anyway, so you will then just need one extra > ethernet socket in the room for all the telephones. > > Michael > Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
Are you manually updating the mySQL tables or do you have a web app. to do that? Robert > Steve Creel wrote: > >>You'll want to #include it. This leaves the burden of the [general] and >>any static configs on sip.conf but allows the script to blindly write out >>from the database to sip_additional.conf >> >>in sip.conf: >>#include sip_additional.conf >> >> >> >>Steve >> >> >> > Excellent, Thanks for that.. I didn't know there was an "include" > command.. > > Do you know if "include" is available in other .conf files eg > extensions.conf?? > > Later.. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be a but rather . Robert > John Brown (CV) wrote: > >>http == hyper text transport protocol >> > So are the entries on your hard drive with a .htm or .html extension not > files? (sorry a little sarcastic I know) *** Big difference beween httProtocol HyperTextMarkupLanguage :-) > >>tftp == trivial FILE trasfer protocol >> >>thus using tftp to do updates seems better. Its also >>a smaller foot print code wise and in boot loader thats >>important. >> > The boot loader size is the the best argument I have heard so far for > using TFTP, but memory is pretty cheap now compared to the days gone by.. > :) > Yes, memory is cheap, disk space is pratically free and processors increase in power every year. But that is not a reason to ignore memory usage or write inefficient programs. IF we used the same programming standards as we had in the last century :-) (70s and 80s) then WinXP would probably run on a 486 with 64MB RAM. >From what I have seen, the Asterisk code must be fairly good. Its running quite nice on my P100, 32 MB system. MusicOnHold ran for 2,5 hours last night without any noticable distortion. Ok.. I don't have many phones hooked up but was fairly surprised that it does as well as it does. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAXTel failing?
> Is anyone else having trouble receiving IAXTel calls? I don't know if > it's my config that's broken or IAXTel that broken. Several people have > given me their IAXTel numbers and calls to them all fail. I can call > FWD numbers via IAXTel just fine. > > --Eric > Eric, I am having a similar problem but am just starting to try and use IAXTEL today for the first time.. Had thought my issue was config but if you had a working config and now its not then maybe its IAXTEL. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH problems
>... > Still: When I call my Asterisk box (which has a fixed IP and is located > within a university network) using X-Lite I get "choppy sound" to say the > least. In fact I can hear only the first half second of what I am > supposed to hear followed by permanent silence. Note that this * box has > no telephony hardware at all. > > Any clues or suggestions what else to try? There is no hardware in > between that could be responsible for silence suppression, but maybe > there is a paramter in Asterisk that I can tune? I tried to use the > "loud" MOH class instead, but it didn't make any difference. :-( > > Cheers, Philipp > When you look in the process list do you see mpg123 processes running. I think there should be 2 for each class you have in the conf file. (at least thats what it seems like on my system. I also copied the mpg123 executable to /usr/bin instead of a link. Not sure if that makes a difference. In theory I would think it would not. With a Grandstream phone and examples from the last days on this list the MOH function when dialed via an extension works absolutly excellent. I've run it for over 2 hours with perfect audio. Gruß aus Friedrichshafen, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gastman crashes on Win32
Can anyone please point me toward the source/binary (linux and Win32) for Gastman?? Robert > Hi, > > The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all > my machines, no error, no log. > Although, the CVS version works great on Linux. > > Is it anybody who knows how to compile it with mingw32 ? Or better, could > anyone, who already has mingw32 installed, make a binary snapshot ? > > Thanks in advance, > > Jean-Christophe > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the X100P a WinModem?
> > > My interrest is radio. I'd like to use Asterisk as a N-way audio > switch between a set of ham radios and to act as a "transcoder" between > a few of the ham-oriented VOIP systems like IRLP, Echo Lnk, Wires and > the like. > > What got me started was one day I was sailing off the coast of > So. Cailornia and had a shirt pocket sized VHF and could talk to > another ham who has riding a bus in England. Voice was being routed > between fixed land based repeters over the Internet. The system is > not easy to use, like say, a cell phone is. I got to thinking > Why Not? and then stumbled on Asterisk while using Google to > find software that could route audio over IP. > > 2 things: First if people want to use Winmodems and build their own hardware, have a blast but don't expect Digium to support it. In the spirit of ham radio this should be no problem. Those that want the installation support get it free if they buy the card so think of the extra $$ as support costs.. That makes them really low cost even at 99,95 USD. I'll gladly pay Digium for the cards in order to know I can call them if there is a problem. Second, check out the app_rpt.c in the ASterisk apps source code (and rpt.conf). I haven't gotten into that yet but it looks like at least a simple interface for a radio. Might be a good place to start. 73, Robert DL4GRC, WB4DHC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI questions..
Jarad, I would be interested in one or 2 of your examples to get an idea of how to get started. Thanks, Robert Friedrichshafen, Germany > On Fri, 2003-10-24 at 05:54, WipeOut wrote: >> First off, can AGI scripts be created using PHP??.. This is where our >> skills are and since PHP can be run from a command line it would be >> easier to create and maintain.. >> > > Yes, you can use PHP just fine for AGI scripting. I recommend, however, > that you use PHP version 4.3.0 or later, due to the updated CLI stuff. > Feel free to contact me off-line if you'd like some examples. > > Jared Smith > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update
> Okay, at the CLi i did a show version and it's still showing the old > version. What I'm attempting to prevent the overwriting of my already > established config files and sound files. Any further suggestions? > When I did the make on Asterisk the first (and only) time, I had to do "make samples" to get the config files. Maybe if you don't "make samples" you won't get any overwrite of the conf files. (no guarantees here since I've never done a CVS Update) You can always employ the long proven method of making a backup of /etc/asterisk and /var/lib/asterisk/sounds before doing the update. I do a backup of /etc/asterisk fairly often anyway in order to snapshop my config. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
>... >> At this moment, Asterisk behind a NAT can't connect to an outside SIP >> provider. If you put asterisk outside your NAT, your inside clients >> can connect to Asterisk and Asterisk will be able to connect to your >> providers. > > I suspected this would be the case. The problem is that I have no control > over the NAT. I guess I'll just have to work on my provider a bit more to > support IAX. > Jonathan, I have the same problem and have solved it by using iaxtel.com. Asterisk talks to IAXtel quite well on inbound and outbound from behind my NAT router. While I don't have the dialplan inside Asterisk completed yet it does do the following: - outbound calls from any internal extension to any service reachable over iaxtel.com. I've tested the following: - USA toll-free numbers (until they stopped working this week.. seems to be an IAXTel problem) - other IAXtel numbers - FWD numbers (1 700 99 x) - inbound calls from FWD to my IAXTEL number ring into the Asterisk box. Currently I play a message then forward them to an internal extension as proof of concept. If you would like the parts of extensions.conf and iax.conf that seem to make it work let me know. I pulled bits and pieces from various places, including a number of the postings on this list over the last 2 days. All of this is rather impressive for me but my wife really wonders if I've lost my sanity... Hunker down everyone.. here comes the solar flare. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nextone softswitch testing and Asterisk long distance
Alexander, I will be happy to help with the testing but since I am behind NAT am not sure it will be of much help to you.. I have 2 Grandstream phones and Asterisk. Robert Friedrichshafen, Germany > Hello All, > > We are looking to test interoperability between Asterisk and Nextone > softswitch. > Please let me know who is wishing to participate. We will open free US > Long distance service for testing. > > Please email me for more details and to be added to testing participants. > To qulaify you need to have already configured Asterisk software > any kind of IP Phone , i.e: SIP IP Phone, H323 Phone, PC2Phone, etc/ > > > Thanks for your time. > > - Alexander > You can also contact by ICQ: 2851311 > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
> My asterisk server(s) are behind NAT, and I am a customer of Vonage > (thrice-over), iconnecthere, and Net2Phone. > > There are still some rough edges (especially with iconnecthere) but > overall it is not correct to say that they won't work. > > B. > Thats great to hear. Can you please share your config files that connect iconnecthere and net2phone via SIP? I think there are a number of people here who have tried and not been able to get it to work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial("SIP/2001-12c8", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
sip.conf is:
[delta3]
type=peer
username=
secret=
host=213.137.73.140
the extension.conf entry is:
exten =>_1706NXX,1,Dial,SIP/[EMAIL PROTECTED]
Am I missing something??
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
> Interesting. Someone thinks that a strategic use for * should be off > this list. Someone thought my FAX modem for * should be off this list. > However, nobody seems to think a 1000 messages about Grandstream phones > should be off this list. > > Personally I would welcome seeing more of what people are doing in the > softswitch area. > > Regards, > Steve > Steve, I agree with you. If the discussion involves * then it should be here. In the case of your fax program I think some people who jumped in after the initial introduction might have thought it was totally separate and didn't make the connection. What I find really good about the fax discussion last week was that in the course of 48 hours it went from a non-working integration to functional in Asterisk. There is a tremendous resource base here... If we aren't interested in a discussion then the delete key or mail filters work wonders. Personally I read at least the beginning of all threads... Never know when a new idea or resource is mentioned. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnecthere connect problem
Hello..
Thanks for the reply.. I'll give this a check later today. Is the first
x in the register command your phone number at ICONNECTHERE? I am
using them with the demo account only as outbound so don't have a phone
number. Maybe this could be the problem.
Regards,
Robert
Friedriedrichshafen, Germany
> Hi!
>
> try to use in sip.conf :
>
> register =>x:[EMAIL PROTECTED]/xx
>
> [iconnect]
> type=friend
> secret=
> username=xxx
> host=sipauth.deltathree.com
> dtmfmode=inband
> context=yourcontext
>
> and in extensions.conf:
>
> exten => _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
>
> This works for me
>
> regards
>
> Miklos
>
>
>
> - Original Message -
> From: "rnc Info Lists" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, October 25, 2003 5:17 PM
> Subject: [Asterisk-Users] Iconnecthere connect problem
>
>
>> I have an Asterisk box behind NAT and am trying to connect to
>> Iconnecthere
>> as was indicated possible earlier. Am getting the following on the
>> Asterisk console:
>>
>> -- Executing Dial("SIP/2001-12c8", "SIP/[EMAIL PROTECTED]") in new
> stack
>> -- Called [EMAIL PROTECTED]
>> == No one is available to answer at this time
>>
>>
>> sip.conf is:
>> [delta3]
>> type=peer
>> username=
>> secret=
>> host=213.137.73.140
>>
>> the extension.conf entry is:
>> exten =>_1706NXX,1,Dial,SIP/[EMAIL PROTECTED]
>>
>> Am I missing something??
>>
>> Robert
>>
>>
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SwissVoice MGCP IP10S
I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
> Citeren rnc Info Lists <[EMAIL PROTECTED]>: > >> I have a SwissVoice IP10S but can not seem to get it to have dialtone or >> dial on *. Calls to or from 3001 don't work. > > Were you able to configure the phones through their webinterface ? > > You could try entering 'mgcp debug' and then power up your phone to see if > it > registers at all... > > > Yes, web config. of the phone works ok. The IP for the Asterisk server is in the call agent field and port 2427. The following comes on the Asterisk console at powerup. The items between the repeat. MGCP Show endpoints doesn't show anything. Evidently the phone isn't registered but not sure why since there doesn't seem to be a place to associate a userid or password. MGCP read: I> RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I> RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart ** from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I> RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
> > Oh, one thought: Did you set your toneconfiguration to Europe or US ? If > you > choose custom you need to configure it another way... > Florian The tone config on the phone is set to Europe. Asterisk is USA.. Hmm.. Will change the phone to USA when I get home and see if that makes a difference. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk FAQ
>>> I've started to write an FAQ for asterisk, available here: >> http://asterisk.pronto.tv/faq.php >> >> Please help me fill it up with the good stuff :) > > Why don't you put it here: > http://www.voip-info.org/tiki-index.php > and folks can updated/edit online? > > > Agreed. There is no need to create YAAWS (Yet Another Asterisk Website. Before you post answers at least please do a check to verify their accuracy. There has been alot of questions about doing SIP from behind NAT. Even this week it was discssed on the list rather extensively that it CAN work but depends on each particular situation. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
> Hi, > >> -Original Message- >> >The portion of extensions.conf is: >> >exten => 3001,1,Dial(MGCP/aaln1,20) >> >> exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) > > Or aaln/1@ should do just fine. However this doesn't explain why there > is no dialtone on the phone.. > > Oh, one thought: Did you set your toneconfiguration to Europe or US ? If > you > choose custom you need to configure it another way... > > Florian > Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message "Waiting for call manager" Additional ideas are appreciated. Will keep plugging away at it. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD connection
As far as I know they do only SIP. If your Asterisk box is behind a NAT firewall then you probably will have problems. > Hi All, > > I have a FWD number and wish to connect it to Asterisk to receive my FWD > calls. > > How I do? > > Is it a register in sip.conf or iax.conf? > > > Regards > > Dave > xmlns:o="urn:schemas-microsoft-com:office:office" > xmlns:w="urn:schemas-microsoft-com:office:word" > xmlns="http://www.w3.org/TR/REC-html40";> > > > > > > > cid:[EMAIL PROTECTED]"> > > > > > > > > mid:Unknown20031101T182611378";> > > style='font-size:12.0pt;color:black;mso-color-alt:windowtext'> > > > > > > > > mid:Unknown20031101T182611378";> > > color=navy face="Times New Roman"> style='font-size:12.0pt;color:navy; > font-weight:bold'> > > style='mso-bidi-font-weight: > normal'> style='font-size: > 12.0pt;color:navy;font-weight:bold'>Registered Office: - 23 First Street, > Low > Moor, Bradford, West Yorkshire, BD12 0JQ. > > style='mso-bidi-font-weight: > normal'> style='font-size: > 12.0pt;color:navy;font-weight:bold'>Company Registration Number: - > 03807643. VAT Registration > Number: > - 734-3363-42 > > style='mso-bidi-font-weight: > normal'> style='font-size: > 12.0pt;color:navy;font-weight:bold'>Telephone / Fax: - 44 (0) 7092 154039. > SIP_Phone: - 1 (747)669 1957 > > style='mso-bidi-font-weight: > normal'> style='font-size: > 12.0pt;color:navy;font-weight:bold'> href="http://www.codepipe.ltd.uk/";>http://www.codepipe.ltd.uk > / http://www.codepipe.com/";>http://www.codepipe.com / E-Mail: > - > [EMAIL PROTECTED] style='mso-bidi-font-weight:normal'> color=navy> style='color:navy;font-weight:bold'> > > > > > > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
> Besides, even if I didn't have the files ready, I wouldn't use my lovely > voice for it - I'll go to a recording studio with a professional (talking > about a production environment) so it's good to know how to do this > yourself, in case the studio doesn't know how to record them in this > format. > For professional recording you can use the same voice as the original prompts.. For details see http://www.digium.com/index.php?menu=thevoice The price seems reasonable to me.. According to John Todd's site the turnaround can be rather fast. (http://www.loligo.com/asterisk/sounds/Sounds-README.txt) http://www.loligo.com/asterisk/ for access to his directory of additional prompts. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
> Hi , > > > I even think to avoid using an installer mainly because the installer > part is bigger that the application himself. > What do you think? > Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the "non-geeks". I only had a short time to run your program last night but it worked well. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Robert Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX softwarephone (for WIndows platform))
> On Mon, 2003-11-03 at 16:27, Alastair Maw wrote: >> On 03/11/03 20:03, Steven Critchfield wrote: >> >> > Sounds like you really need a C programmer and get into the guts >> > of asterisk. Can't get more flexible than having the source code >> > yourself to do anything you want. You could add your DSP routines into >> > the dsp.c file and call them when needed. You can also write a >> asterisk >> > application and have direct access to all the audio in every direction >> > just as you want it. >> >> But C isn't as maintainable as nice Java apps, and it's as simple as >> that. Basically, I'm after the most powerful interface possible to >> Asterisk, but trying to make it as friendly as possible to code things >> against. As far as our organization is concerned, that pretty much means >> Java objects. > > So you bought that line of Marketecture didn't you. I think there are > several large open source projects that prove that C is maintainable. > Maintainability is really a function of organization. If you can't be > organized, you will not produce very maintainable C code. > > I'll point out that I am not a C programmer, but making patches to > asterisk isn't that difficult. I have also made patches to the kernel > without too much hair pulling. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > Steve, You are right... Lots of proof that C is maintainable. I don't profess to be a C, VB or JAVA expert but have programmed for longer than I care to admit. What matters most is good solid and tight code regardless of the language. It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. Asterisk does rather well on my Pentium 100/32 MB RAM. Wish I still had the Pentium 75 to try it on. It must really boogy on the bigger boxes. I contend that the "most powerful interface" is one that meets the requirements of the customer (1st requirement), is written to be the most efficient (2nd requirement) and maintainable (3rd requirement) as possible. The language to be used is the selection of the person doing the development. I'm not a fan of any Microsoft product but they do have a place in the world (for now). Kudos to Dan for his IAX phone. It works. He is responsive to bug fixes. Hopefully he will continue the development. Mark's offer of direct help I think speaks volumes about the importance of GPL IAX softphones for Win32. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
> Daniel, > > the MGCP log you sent shows you sending the digits and asterisk receiving > them, however after that either nothing happens (infinite digittimeout) or > you cut the log short. Can you also send some console output with 'mgcp no > debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be > usefull as well ? > > Also, can you tell us your phone's firmware ? (the IP10) > > I had one minor issue with the IP10 because of an older firmware version, > a > simple upgrade resolved it (by the way, in my case it was interpreting > digits twice in some cases, i.e. dialling 326 would make asterisk think I > was calling 33226) > > Best regards, > Florian > FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP10S and Handset
With some lively IP10S discussions here maybe someone knows about this issue: I can use the speaker phone ok. However the handset and switch hook do not seem to work. If I enable "headset" then I can get audio via the handset but still have to use the speaker phone button to take ot "off hook". Seems a bit wierd.. I have sent it to Swissvoice but no answer back yet. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP10S and Handset
> Hi Robert, > > I haven't the HeadSet model but the lan switch model so I can't be of > any help for you. > > Daniel > I have the IP10S LAN Switch model too.. Thats why I find it wierd that the headset setting makes the difference ! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
> Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net > > I've posted my demp weather report Asterisk AGI script at > http://www.fnords.org/~eric/asterisk/downloads/ > Eric, Can you comment on the difference in installation ease for Festival and Cepstral? Regards, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
> Cepestral was installed and working within 10 mins of my decision to > purchase it. It's $30.00 and can be purchased on their web site and > they give you a download. They have a demp on their website that will > do text-to-speech and give you a .wav file to download and listen to. > Download, unpack it, run their install.sh, answer a couple of questions, > read the man page and you're done. > > With Festival I had to figure out exactly which tarballs to download > (there was a total of 18 tarballs to download if you count all the > Festival voices plus the MBROLA voices), then I had to figure out how to > install Festival, then MBROLA, I never have figured out how to actually > INSTALL festival, I just run it out of the source directory. It's very > picky about paths and such. > > I'm not a big fan of commercial software. For TTS most of the software > either is Windows only or costs several thousand dollars (and sometimes > both). If it's a choice between spending two thousand for something > like Rhetorical TTS or using Festival, I'll pick Festival. If it's a > choice between spending thirty dollars for a TTS system or using > Festival, I'll happily spend the $30. > Thats a very easy ROI since one hour of a technical resource to setup Festival is easily double the 30 USD. Maybe the Cepestral folks have figured out that making a little money from alot of people will be much better than alot from only a few. I'll buy Cepestral and skip the pizza on Friday night. Net result will be about break even ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail RFC
Earlier today someone posted a RFC number related to voice mail. Unfortunatly I deleted the message so have lost the number and don't see it yet in Google. Can you please resend that to me? Thanks, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
> > I think it is time to start commercial Pro version (not expensive !!!) of > Asterisk. > In my company we already made decision to do it, to offer people > ready-to-go solution. But is is hard to do anykind of such product without > Digium and Mark's support. > Mark I think you are very overloaded with all projects, maybe we can > help > with Asterisk project. "Asterisk Basic" will stay as it is now, but we > will > be developing > "Asterisk Pro". Correct me if I am wrong, but unless you have a license from Digium directly then you must sell your "Pro" version software under GPL. What you do for documentation/packaging is probalby not covered under GPL. You make some good points but I think that the solution is not to commercialize everything. There is starting to be a trend of businesses (and governments) turning away from commercialization (ever so slowly but it is in that direction). Pick something that is missing and contribute that to the community. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 1
> Is anyone running Asterisk under Fedora Core 1 > (http://fedora.redhat.com/)? > If so, did everything with Asterisk work properly? I'm looking to migrate > from Red Hat 8.0 to Fedora this week. > > Thanks. > Interesting question... Since RedHat will in the future have only their Enterprise version I wonder if Digium/Mark will develop running on Fedora or move to some other system. Guess the jury is still out on what Novell will do with the SUSE distribution long term. Hopefully they maintain the current distribution package scheme. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone-101 & MWI
Max, That is what worked for me. if you want the MESSAGE button on the GS to dial the VM then put whatever extension you have defined for VM in the field "Voice Mail UserID" via the GS Admin Web Interface. Robert > > Hi Folks, > > Bit of a newbee here, so please be gentle. :) > > I'm trying to get the message waiting indication working on a > budgetone-101. Is it as simple as putting `mailbox=n' where n is the > mailbox number into sip.conf? > > Is there anything else I should check or set? > > -Cheers Max. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EU SIP Phone providers
Does anyone know of SIP phone providers (Grandstream in particular) who are located in Germany (or the EU) Thanks for any info. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background only responds to 1 digit
I have a problem where the Background application only seems to work if one digit is pressed. Extensions with multiple digits just timeout and asterisk hangs up. Below is the relevant excerpt from extensions.conf. In this example, pressing 2 will access the service menu. Then pressing 1 will do the echo test ok but pressing 8463 or 33 will cause an invalid extension message. Any ideas for a solution are appreciated. Robert [default] exten=> s,1,ResponseTimeout,10 exten=> s,2,Background(rnc-mainmenu) exten=> 1,1,Goto(local-extensions,2001,1) exten=> 2,1,Goto(services,s,1) [services] exten=> s,1,ResponseTimeout,10 exten=> s,2,Background(rnc-svcmenu) exten => 1,1,Answer exten => 1,2,Playback(demo-echotest) exten => 1,3,Echo() exten => 1,4,Playback(demo-echodone) exten => 1,5,Wait(1) exten => 1,6,Playback(vm-goodbye) exten => 1,7,Wait(1) exten => 1,8,Hangup exten =>33,1,Answer exten =>33,2,MusicOnHold(random) exten =>8463,1,DateTime() exten =>8463,2,Wait(2) exten =>8463,3,Playback(vm-goodbye) exten =>8463,4,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Business discussion again
> Why don't we just add it on the DIgium list server, wouldn't that make > more sense, to have a single place for all list memberships? > > Mark > OR even just leave the discussion on asterisk-users... If we create new lists everytime some people disagree with a topic being on-list then we will have not 2 or 3 lists but many more. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Are you also able to make outgoing calls via Iconnecthere? If so do you mind posting your config? I tried their 10 minute trial a couple of months ago but was not able to get a connection. Thanks, Robert > I'm receiving calls on my asterisk server from iconnecthere. My asterisk > server is behind nat but it still seems to be working fine. > AJ > > On Fri, 21 Nov 2003, Chris HARIGA wrote: > >> Hi, >> >> Is anyone using the iconnect on Asterisk to receive and to place calls? >> >> Best regards, >> >> Chris HARIGA >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
