Re: [asterisk-users] Display channels and codecs
Try sip show channels from the CLI - Original Message - From: Scott Moseman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 12, 2007 6:12 PM Subject: [asterisk-users] Display channels and codecs Is there an easy way to show all active channels AND the codecs they're using? Other than going through EACH channel individually to check the codec which is, obviously, not a very efficient process. Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI with System() ?
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote: Hi. You mean to use the AGI funtion in the particular programming language? yeah. i tried, same results.. : T I guess that this is a permissions issue. Check what you get in the standard error. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware requirements
I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Olivier wrote: I was told yesterday (by Cantata guy) that T.38 demands a good level of QoS. That surprised me a lot as I thought the whole purpose of T.38 was to avoid SIP and ToIP latency. T.37 is the answer to reliability, but most people don't want to use it for totally stupid reasons. T.38 is a fudge to make real time FAX over IP less flaky. It isn't all that robust, it just isn't as awful as FAX over VoIP. Another editor (Interstar) told me T.38 passthrough doesn't work. That's not true. As long as the passthrough has fairly low latency (and, of course, a solid reliable implementation), it shouldn't impact the results. As devil lies in details and I couldn't get any, I'm not sure these words would be of any use. In a sane world all FAX would have been T.37 from a few months after that spec was released. :-) Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
On Sun, 14 Oct 2007, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. Look more. There are 100's of pages on it. Start at http://www.voip-info.org/wiki/ What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? You would get away with a 1GHz intel (or intel like) processor for this system, so the answer is: Any modern server will do the job you need it to. Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? Can't help you there I'm afraid. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.13 build crashed
Alan Lord wrote: As I said, 1.4.12 builds fine. I'll do a bit more digging and if I find a cause I'll report it upstream. I started to investigate this problem, but it seems it was something to with me rather than the build process... Hi Thanks for your effots, Please ignore this - it was my bad I think I've just re-built the stock 1.4.13 just to make sure, and it built fine. The only thing I can think of that 'might' have been the cause was that I had just upgraded my kernel to the 2.6.23 release but still - at that time - only had zaptel modules that were built against my previous 2.6.22.9 kernel. This time I have now got zaptel modules built against the later kernel. Strange. Because the build of Asterisk only involves zaptel.h, the userspace interface. Anyway sorry for the noise - I have just successfully built 1.4.13 on my LFS system. GCC-4.2.1, glibc-2.5.1, kernel-2.6.23. Good to know. I sugest you reply accordingly on the list. At least for the benefit of others who will run into your problem in a web search... Cheers, Tzafrir Cohen Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get egress SIP call Id
Hi Ray, Am Dienstag, den 09.10.2007, 10:10 -0500 schrieb Ray Chen: Hi Philipp, Thank you for your response to my question. I am working on a project which uses Asterisk as the voice engine. I need to get the ingress and egress sip call id for a call to write call CDR. (Asterisk CDR does not meet our customer requirments). If there is no any easy way to get it I might need to create a seperate process/thread to query manager interface as you mentioned. Thanks you, Maybe You can do the trick with a Dead-AGI script. Run that script in the 'h' priority and set the Userdata field of the CDR. Than configure Your CDR to include the userdata field in the output (depends on Your CDR backend). I have not tested this, but it might be easier than hacking a new process... For the details look at the description of the * CLI: (core) show function CDR and the description of AGI: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AGI HTH, Karsten ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI with System() ?
Ok, this is what worked: EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm the -rf eliminates the hassle.. a dream come true it worked ! On 10/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote: Hi. You mean to use the AGI funtion in the particular programming language? yeah. i tried, same results.. : T I guess that this is a permissions issue. Check what you get in the standard error. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI with System() ?
On Sun, Oct 14, 2007 at 05:43:27AM -0700, Dominic Son wrote: Ok, this is what worked: EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm the -rf eliminates the hassle.. a dream come true it worked ! -r sure wasn't needed . -f then? But this is the default of rm. The shell got in your way? EXEC System /bin/rm /var/lib/asterisk/sounds/blah.gsm And still, getting a System through Asterisk is an overkill and introduces an extra layer of complication and inefficiency. Use you language's equivalent of system(3) . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Hi All, I've been trying to send a message to the list for the past 3 days, but I neither get bounces nor the message appearing in the list, so someone on IRC sugested I reply to an existing message. My subject is related to this message, although slightly different. Apologies if my actual messages appear in the list. Here's a paste of one of my past messages: -- I'm playing with a PA-type setup, where people can dial a number, and Asterisk would place a call file to get another phone to dial in (auto answering) and play to it a sound. It's woking, but I'm getting some errors, as I'll paste below. So, my setup: Asterisk 1.4.13 Debian GNU/Linux 4.0 Linux Kernel 2.6.18-5-686 SIP client: snom360 5.3 soft-phone SIP/[EMAIL PROTECTED] My call file: Channel: Local/[EMAIL PROTECTED]/n Extension: 6600 extensions.conf: [from-sip] exten = 6600,1,Answer exten = 6600,n,Wait(1) exten = 6600,n,Playback(demo-thanks) exten = 6600,n,Hangup [localtest] exten = pa,1,SIPAddHeader(Call-Info:sip:asterisk\;answer-after=0) exten = pa,n,Dial(SIP/pa) The Console (-rvvv): -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1) -- Executing [EMAIL PROTECTED]:1] SIPAddHeader(Local/[EMAIL PROTECTED],2, Call-Info:sip:asterisk;answer-after=0) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/pa) in new stack -- Called pa -- SIP/pa-081ddd30 is ringing -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2 == Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so falling back to exten 's' -- Executing [EMAIL PROTECTED]:1] Wait(Local/[EMAIL PROTECTED],1, 1) in new stack [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin . . . And it keeps going, really fast, indefinately, until hung up. Any ideas? How about the 's' error above? Thanks, bu Nick Couchman wrote: Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Actually, forget everything else. Even when I simply pick up the handset and dial 6600, I get those errors in console, so it's not related to paging or call files or anything special, I guess.. Any ideas? bu [EMAIL PROTECTED] wrote: Hi All, I've been trying to send a message to the list for the past 3 days, but I neither get bounces nor the message appearing in the list, so someone on IRC sugested I reply to an existing message. My subject is related to this message, although slightly different. Apologies if my actual messages appear in the list. Here's a paste of one of my past messages: -- I'm playing with a PA-type setup, where people can dial a number, and Asterisk would place a call file to get another phone to dial in (auto answering) and play to it a sound. It's woking, but I'm getting some errors, as I'll paste below. So, my setup: Asterisk 1.4.13 Debian GNU/Linux 4.0 Linux Kernel 2.6.18-5-686 SIP client: snom360 5.3 soft-phone SIP/[EMAIL PROTECTED] My call file: Channel: Local/[EMAIL PROTECTED]/n Extension: 6600 extensions.conf: [from-sip] exten = 6600,1,Answer exten = 6600,n,Wait(1) exten = 6600,n,Playback(demo-thanks) exten = 6600,n,Hangup [localtest] exten = pa,1,SIPAddHeader(Call-Info:sip:asterisk\;answer-after=0) exten = pa,n,Dial(SIP/pa) The Console (-rvvv): -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1) -- Executing [EMAIL PROTECTED]:1] SIPAddHeader(Local/[EMAIL PROTECTED],2, Call-Info:sip:asterisk;answer-after=0) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/pa) in new stack -- Called pa -- SIP/pa-081ddd30 is ringing -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2 == Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so falling back to exten 's' -- Executing [EMAIL PROTECTED]:1] Wait(Local/[EMAIL PROTECTED],1, 1) in new stack [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin . . . And it keeps going, really fast, indefinately, until hung up. Any ideas? How about the 's' error above? Thanks, bu Nick Couchman wrote: Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with sendinf to list. Was: Re: Paging in Asterisk
On Mo, 15 Okt 2007, [EMAIL PROTECTED] wrote: I've been trying to send a message to the list for the past 3 days, but I neither get bounces nor the message appearing in the list, so someone on IRC sugested I reply to an existing message. Same with me here! -- Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0 http://wwwkeys.de.pgp.net/pks/lookup?op=getsearch=0x7E354E4D5DD5D0E0 signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote voicemail in two Asterisk
http://en.wikipedia.org/wiki/Network_File_System_(protocol) On 10/12/07, Pepo [EMAIL PROTECTED] wrote: Using two Asterisk connected between they, How do I can check the voicemail in a remote system but working like *97? I mean dont want ask the voicemail box, just the password and go to the voicemail of caller. If I have the same extensions in the two Asterisk it doesn't work. Thanks. -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Start' in extension rules
Tilghman Lesher wrote:
On Saturday 13 October 2007 18:08:49 Turbo Fredriksson wrote:
Quoting Philipp Kempgen [EMAIL PROTECTED]:
exten = s,1,Answer()
exten = s,n,Goto(s-${DIALSTATUS},1)
This still doesn't make sense because you did not Dial()
before jumping based on ${DIALSTATUS}.
Ok, make sense. But still no go:
- s n i p -
[default]
exten = s,1,Answer()
exten = s,2,Dial(SIP/${EXTEN},20,t)
exten = s,4,Goto(default,s-${DIALSTATUS},1)
1, 2, 4...
Also, the s extension is only executed under a few situations. If a
call comes in on an FXO port or you execute a macro or you have
immediate=yes for that channel.
s should not stand for start. It should be called stupid as in
the device is too stupid to tell Asterisk what the dialed extension is.
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Re: [asterisk-users] Display channels and codecs
The Management Interface has an Action: Status that sends back all active channels. No codecs, but it's a start. Girts On 10/13/07, Dovid B [EMAIL PROTECTED] wrote: Try sip show channels from the CLI - Original Message - From: Scott Moseman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 12, 2007 6:12 PM Subject: [asterisk-users] Display channels and codecs Is there an easy way to show all active channels AND the codecs they're using? Other than going through EACH channel individually to check the codec which is, obviously, not a very efficient process. Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] difference between FXO interfaces !
Hello everybody, Which one is a better choice 1. Gateway device with FXO - SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP. Office is having 12 phone lines. Thanks in Advance to all who shared his/her wisdom. -- With Regards, Mandeep Singh Bhabha email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with sendinf to list. Was: Re: Paging in Asterisk
Volker Sauer wrote: On Mo, 15 Okt 2007, [EMAIL PROTECTED] wrote: I've been trying to send a message to the list for the past 3 days, but I neither get bounces nor the message appearing in the list, so someone on IRC sugested I reply to an existing message. Same with me here! Yep, me too. Ron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR
Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was there when the same situation happened in 1.4.4. I also have this message in the console when an outgoing, noanswered call terminates: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/2-1' not posted What cause this behavior? Is it a bug or misconfiguration. I tried to google this issue but unfortunately it does not reveal anything useful. Any help would be gladly expected. Cheers Andrew ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Hello all, I'd like to thank everyone's input which I'll sumarize and comment on bellow. As in all complex solutions, there are no quick answers and no 100% correct solutions. There are trade-offs to be made among very different possiblities... Of course, the purpose of my original post was exactly get some feedback on what I initially designed and to widen my perspective on the particular subject by hearing different approaches to the problem. It's been great ! :-) For those interested, here is the summary: 1. from Mojo with Horan Company, LLC sheet-fed PDF scanners - desktop PDF - print to HylaFAX good: - nice idea, makes use of centralized HylaFAX server bad: - needs investment in replacing current equipment not sure if: - FAX users are PC savvy - there is a PC near every FAX 2. from Andreas van dem Helge suggests using a T.38 fax provider good: - would offload the gatewaying to a provider need to know: - whether T.38 is effectively solid under such scenario (see last comment, below) he also comments: - no success with callweaver T.38 gateway with some betas (answer to his question: the channel banks allow for the connection of analog FAX machines to the asterisk servers via PRI) - then says the topology I presented has too many PRIs: PSTN --PRI-- ast 1.2 --PRI-- AS5300 --SIP-- T.38 ATA he suggests something I don't quite understand (are these three parallel flows ? or does it represent one PRI going to a single AS5300 which would deliver the calls to T.38 ATAs or asterisk based on DDI ? what's the difference between the last two lines, can the AS5300 talk SIP/T.38 directly to an ATA without a SIP proxy ?): PSTN --PRI-- AS5300 --SIP-- ast 1.2 PSTN --PRI-- AS5300 --SIP-- ast 1.4 --SIP-- T.38 ATA PSTN --PRI-- AS5300 --SIP-- T.38 ATA 3. from Olivier shares information he got from Cantata where T.38 requires good levels of QoS my comment: I though T.38 was created to bypass those types of technical hurdles -- interesting ! (as I'll note below, Steve Underwood helps clarifying this notion) 4. from Phillip von Klitzing suggests that some bigger MFC printer/copy/fax combos can do FAX via SMTP good: - great, if it's over SMTP it'll work bad: - small offices won't justify such a big investment (I used to work for HP, I know how much those beasts can cost!) ;-) ...unless anyone's aware of a small FAX machine that can do SMTP ! (btw, there are some sheet-fed network scanners that can do SMTP -- see first comment) he also recalls an important issue: are you sure you want to rely 100% on IP only in your sattelite offices ? It might be wise to have 1 (analog?) line installed anway great point -- this has always been a possibility in the back of my mind... the only thing we'd loose in a setup where the remote office FAXes are directly attached to local analog lines is the ability to do integrated CDR processing for those FAX usages 5. from Benny Amorsen reminds that those big MFC boxes require the fax as email address for sending -- maybe too complex in day to day usage ? how tech savvy are the users ? another good point -- apart from their cost, in terms of usability, they might come short... or be too complex for someone with basic FAX machine abilities 6. from Steve Underwood reminds that T.37 (store and forward instead of realtime) is the answer to reliability... T.38 isn't all that robust, it just isn't as awful as FAX over VoIP he then concludes In a sane world all FAX would have been T.37 from a few months after the spec was released great info -- so, where is the T.37 compliant equipment ? (gateways, ATAs, FAX machines ?) Again, thanks a lot for the feedback (keep those posts coming!). Meanwhile I'll move on to further investigate some of the alternatives you proposed. Cheers, -- exvito ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
Hello Andrew, In order to help you, could you please provide your dialplan ? BR Mathieu Andrew Nowrot a écrit : Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 http://1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was there when the same situation happened in 1.4.4. I also have this message in the console when an outgoing, noanswered call terminates: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/2-1' not posted What cause this behavior? Is it a bug or misconfiguration. I tried to google this issue but unfortunately it does not reveal anything useful. Any help would be gladly expected. Cheers Andrew ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
On 10/14/07, Andrew Nowrot [EMAIL PROTECTED] wrote: Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was there when the same situation happened in 1.4.4. I also have this message in the console when an outgoing, noanswered call terminates: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/2-1' not posted What cause this behavior? Is it a bug or misconfiguration. I tried to google this issue but unfortunately it does not reveal anything useful. Yes, there's a change. For me it's completely unacceptable, so i reverted the patch (http://bugs.digium.com/view.php?id=10659). The thing is that one-channel CDRs without answer are not written. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResponseTimeOut() and t extension
Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why? Is it because I am originating the call from local extension (an handset connected to FXS port) and the call should be originated from FXO or IP Trunk, or what is the problem exactly? [Test_Bilal] include = KuwaitInternal include = EgyptInternal exten = 1000,1,Goto(s,1) exten = s,1,Answer() exten = s,2,ResponseTimeout(5) exten = s,3,Background(WelcomeMessage) exten = 0,1,Dial(SIP/EgyptOperatorSIP,10) exten = 0,2,Background(WelcomeMessage) exten = 0,2,Playback(vm-nobodyavail) exten = 0,3,Hangup() exten = 0,102,Playback(tt-allbusy) exten = 0,103,Hangup() exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(EgyptIncomingPSTN,s,1) exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup() Any help?? Regards Bilal Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
bilal ghayyad wrote: Hi List; Can someone advise me why in the below context, it You never told us what version you are running. If it's version 1.2, make sure you have set priorityjumping=no in your extensions.conf or use the waitexten application. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Does 'sip show peers' actually show the phone as registered? PaulH On Mon, 2007-10-15 at 02:05 +1000, [EMAIL PROTECTED] wrote: Actually, forget everything else. Even when I simply pick up the handset and dial 6600, I get those errors in console, so it's not related to paging or call files or anything special, I guess.. Any ideas? bu [EMAIL PROTECTED] wrote: Hi All, I've been trying to send a message to the list for the past 3 days, but I neither get bounces nor the message appearing in the list, so someone on IRC sugested I reply to an existing message. My subject is related to this message, although slightly different. Apologies if my actual messages appear in the list. Here's a paste of one of my past messages: -- I'm playing with a PA-type setup, where people can dial a number, and Asterisk would place a call file to get another phone to dial in (auto answering) and play to it a sound. It's woking, but I'm getting some errors, as I'll paste below. So, my setup: Asterisk 1.4.13 Debian GNU/Linux 4.0 Linux Kernel 2.6.18-5-686 SIP client: snom360 5.3 soft-phone SIP/[EMAIL PROTECTED] My call file: Channel: Local/[EMAIL PROTECTED]/n Extension: 6600 extensions.conf: [from-sip] exten = 6600,1,Answer exten = 6600,n,Wait(1) exten = 6600,n,Playback(demo-thanks) exten = 6600,n,Hangup [localtest] exten = pa,1,SIPAddHeader(Call-Info:sip:asterisk\;answer-after=0) exten = pa,n,Dial(SIP/pa) The Console (-rvvv): -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1) -- Executing [EMAIL PROTECTED]:1] SIPAddHeader(Local/[EMAIL PROTECTED],2, Call-Info:sip:asterisk;answer-after=0) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/pa) in new stack -- Called pa -- SIP/pa-081ddd30 is ringing -- SIP/pa-081ddd30 answered Local/[EMAIL PROTECTED],2 == Starting Local/[EMAIL PROTECTED],1 at default,6600,1 failed so falling back to exten 's' -- Executing [EMAIL PROTECTED]:1] Wait(Local/[EMAIL PROTECTED],1, 1) in new stack [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for gsmtolin . . . And it keeps going, really fast, indefinately, until hung up. Any ideas? How about the 's' error above? Thanks, bu Nick Couchman wrote: Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
On Sunday 14 October 2007 17:35:04 bilal ghayyad wrote: Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why? Is it because I am originating the call from local extension (an handset connected to FXS port) and the call should be originated from FXO or IP Trunk, or what is the problem exactly? [Test_Bilal] include = KuwaitInternal include = EgyptInternal exten = 1000,1,Goto(s,1) exten = s,1,Answer() exten = s,2,ResponseTimeout(5) exten = s,3,Background(WelcomeMessage) exten = 0,1,Dial(SIP/EgyptOperatorSIP,10) exten = 0,2,Background(WelcomeMessage) exten = 0,2,Playback(vm-nobodyavail) exten = 0,3,Hangup() exten = 0,102,Playback(tt-allbusy) exten = 0,103,Hangup() exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(EgyptIncomingPSTN,s,1) exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup() Go read the top of configs/extensions.conf.sample, specifically the part about the autofallthrough parameter. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
bilal ghayyad wrote: Can someone advise me why in the below context, it does not run the Background step? [Test_Bilal] include = KuwaitInternal include = EgyptInternal exten = 1000,1,Goto(s,1) exten = s,1,Answer() exten = s,2,ResponseTimeout(5) exten = s,3,Background(WelcomeMessage) exten = 0,1,Dial(SIP/EgyptOperatorSIP,10) exten = 0,2,Background(WelcomeMessage) exten = 0,2,Playback(vm-nobodyavail) exten = 0,3,Hangup() 1, 2, 2, 3. - That is not supposed to work. exten = 0,102,Playback(tt-allbusy) exten = 0,103,Hangup() Don't use priority jumping. There are many examples how to do it better. btw: Kuwait, Egypt - Are you going to become a VoIP provider ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skills Based Routing
Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The other alternative is to write our own queue system with AGI, effort++ though :-) TIA. Cheers, Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skills Based Routing
nick, I am actually playing with skills based routing right now... how would you propose to send multiple calls requiring different skills into a single queue and have agents w/o that particular skill in the same queue? daveC Nick Brown wrote: Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The other alternative is to write our own queue system with AGI, effort++ though :-) TIA. Cheers, Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AA50 Paging
Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Kelly opal wrote: I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Can't help you with this, but do you mean it reboots/crashes the machine? Or does it restart asterisk? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
I’m not sure if this will work on the Grandstream phones but I use this for the Linksys phones. exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = ,n,Dial(SIP/201) exten = ,n,HangUp I would guess it would work with multiple phones, i.e., exten = ,n,Dial(SIP/201 SIP/202 SIP/203 SIP/204) You may need to check the phone is configured for paging auto answer. The Linksys has a field of Paging Serv and is set to yes. Let me know if it works. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Monday, 15 October 2007 7:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi, I am curious. What version of asterisk is running on that AA50? Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Sunday, October 14, 2007 5:46 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFC/R2 protocol varient - sri lanka/Nortel DMS 100
Hi All, We successfully installed MFC/R2, chan_unicall.so with asterisk ver 1.2.6. asterisk is loading properly and we can see US show channels working fine. We are using digium Te120P card. Now we are trying to setup E1 link with Nortel DMS 100, which is resides at one of telco provider in Sri Lanka. But we don't know what is the exact protocol varient to use. Is anyone help us out on this reagard. How do we know the exact details of the protocol varient we have to use?. Thanks Regards, Vidura Senadeera, Senior solutions Specialist, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi It restarts asterisk. The unit does not reboot. Kelly On Mon, 2007-10-15 at 04:22 +0200, Philipp Kempgen wrote: Kelly opal wrote: I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Can't help you with this, but do you mean it reboots/crashes the machine? Or does it restart asterisk? Regards, Philipp Kempgen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi I tried that. Unfortunately it is the Dial command. The first phone to answer wins and the rest are dropped from the channel. Thanks Kelly On Mon, 2007-10-15 at 12:25 +1000, Klaverstyn, David C wrote: I’m not sure if this will work on the Grandstream phones but I use this for the Linksys phones. exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = ,n,Dial(SIP/201) exten = ,n,HangUp I would guess it would work with multiple phones, i.e., exten = ,n,Dial(SIP/201 SIP/202 SIP/203 SIP/204) You may need to check the phone is configured for paging auto answer. The Linksys has a field of Paging Serv and is set to yes. Let me know if it works. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Monday, 15 October 2007 7:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi Digium support says it is built on the 1.4 platform. Kelly On Sun, 2007-10-14 at 22:28 -0400, Joseph Begumisa wrote: Hi, I am curious. What version of asterisk is running on that AA50? Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Sunday, October 14, 2007 5:46 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
About memory, I think 512MB will be more than enougth. And hard drive requirements depends on the configuration of your voice boxes, but any modern server will be OK, I don't think that you need more than 20GB... On 10/14/07, Paul Hales [EMAIL PROTECTED] wrote: We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between FXO interfaces !
The model AP200 that you are giving as example is 2 port only... and i'm not sure about the price... I know that codec conversion is one of the most cpu-intensive task that asterisk has to do, so, you can chose a Digium/Sangoma card with a powerful server doing the work or you can also use a VoIP gateway with a cheaper and less powerful asterisk box. It depens so much in your resources.. On 10/14/07, Mandeep Singh Bhabha [EMAIL PROTECTED] wrote: Hello everybody, Which one is a better choice 1. Gateway device with FXO - SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP. Office is having 12 phone lines. Thanks in Advance to all who shared his/her wisdom. -- With Regards, Mandeep Singh Bhabha email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
20GB should be fine - unless you want to do a lot of recording. PaulH On Sun, 2007-10-14 at 21:07 -0600, Edgar Guadamuz wrote: About memory, I think 512MB will be more than enougth. And hard drive requirements depends on the configuration of your voice boxes, but any modern server will be OK, I don't think that you need more than 20GB... On 10/14/07, Paul Hales [EMAIL PROTECTED] wrote: We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel.c switches to gsm even when sip.conf only allows ulaw
Hi Guys, I have noticed a weird behavior in 1.4.12. When using Authenticate or DISA in the dial plan the channel immediately switches to gsm format (if you request a password) or slim (if you run DISA without password). The debug log says... === [Oct 14 21:23:00] DEBUG[9013] channel.c: Set channel SIP/1970xx-0821aad0 to write format gsm [Oct 14 21:23:00] DEBUG[9013] rtp.c: Difference is 82008, ms is 10271 === It does this without caring about the fact that you are ONLY allowing ulaw in the channel configuration. I have so far played with SIP but it seems the behavior is there for other channels as well (briefly tried it on IAX as well) The problem with this is that some SIP providers (ViaTalk) only allows DTMF of the type inband, which only works on ulaw. Therefore this switch to GSM makes it impossible to enter the DISA or Authenticate password. This behavior seems to have been introduced with 1.4.12 as I didn't have any problem in 1.4.11. Has somebody else seen this. Cheers, // Jonas ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
