[Asterisk-Users] FAX over SIP
Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line - mp108fxo - Asterisk -- mp108fxs --- fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid extension ff in context ). 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call to this extension but when the fax answers the call is dropped ( don't have here the SIP debug output ) but seems that when * tries to make the bridge the mp108 fxo sends a BYE. 3) I dialed in to * with a phone ( external line and internal extension ) and dial extension for the fax and i cann hear the fax, so the call is not dropped, the bridge is established successfully. 4) If I pickup the call on the fax machine ( it has a phone set ) and then pressed the 'start' button to start de fax receiver, then, the two faxes talked to each other and the fax is received well. Seems that the problem is only when the fax answer automatically ( could be the tones the receiving fax plays ? ), the same problem happens when i try to use hylafax to receive the fax. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 Users
I am aware of a least a few people (including me) who were using the Packet8 service along with Asterisk for outgoing calls. Last night Packet8 did a software upgrade and both last night and this morning I have been unable to make any outgoing calls. Has anyone else noticed this behavior and/or been able to correct it? I get Got SIP response 403 Forbidden back from 4.42.235.170 when trying to make calls. This is a wild guess, but let me throw it out there. Take a look at the User-Agent: header from a Cisco ATA-186 or maybe from one of the Packet8 devices. Then, go into chan_sip.c and alter the User-Agent: setting from Asterisk PBX to one of those values. See if your dialing works. Other providers have been known to implement this type of User-Agent: filter to (crudely) try to disallow certain types of equipment not on their authorized lists from accessing their systems. I don't know if that's what happened with Packet8, but it's worth a shot, and will probably tell you more than calling their help desk. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP on TCP
Abi- I would suggest you then put in a feature request into http://bugs.digium.com/ and include this list thread. Perhaps, in their copious spare time ;) the Asterisk development folks (or you, or someone else) might create a patch to handle SIP TCP connections. Of course, since Asterisk is open source, feel free to create the patch, sign the disclaimer (on the bugs site) and submit it! Everyone wins. I have not personally found carriers that insist on TCP signalling for SIP, but I suppose I can see why (despite the higher load on core SIP proxies.) If you are unable to make Asterisk work in a real business environment as you describe, then this is probably a serious issue that should be considered as a more important feature request than others. I don't know how to automatically determine TCP or UDP for outbound connections without blocking or putting in some really nasty UDP failure detection modes. Maybe this would best be configured to start with just protocol=[tcp,udp] as the only options, but I leave that to the patch writers, as they know far better than me how to make that work. JT JT, We use 2 providers iPCB.NET and NTT (backup) and both require signalling on TCP only. Interestingly, I find this to be the norm amongst Cisco powered providers. As * marches on to the #1 telco product and SIP to the #1 protocol of choice, protocol=[tcp,udp,auto] feature is a good idea in sip.conf. I will add it as a feature. Master -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, 4 September 2003 3:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP on TCP Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master Out of curiosity, what SIP provider is that? I've never seen any SIP providers that even support SIP over TCP, much less mandate it. If that is required, maybe a protocol=[tcp,udp,auto] feature is a good idea in sip.conf. JT ___ 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] Still no audio on SIP phone
Whould you mind sharing the important bits of your X-lite config with me First, let me explain that I have only used x-lite to do a few quick tests. I verified audio in both directions and dtmf to work ok. The following are the settings I changed, with the exception of voicemail.conf and extensions.conf if you want to receive calls and have voice mail work on the sip phone. I have asterisk running at 10.0.0.2 and x-lite phone running at 10.0.0.8 In /etc/asterisk/sip.conf I have the following: ;X-Lite www.xten.com [xlite] callerid=CALLER NAME (555) 555- type=friend nat=no secret=password host=dynamic defaultip=10.0.0.8 mailbox=55 canreinvite=no dtmfmode=rfc2833 (network, sip settings)? Did you download the generic X-lite or the one for FWD? Then in the generic phone config xlite v2.0 for windows: http://brands.xten.net/x-lite/download/X-Lite_Install.exe http://www.xten.com/docs/X-PRO_v2_Manual.pdf click menu button system settings sip proxy [default]: 10.0.0.2 Enabled:yes Display Name: CALLER NAME User Name: xlite Password: password Domain/Realm: 10.0.0.2 SIP Proxy: 10.0.0.2:5060 Send Internal IP: On ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming CallerID management
Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a different welcome message. If a customer calls the 06x01 - it should be answered by the message for company A If a customer calls the 06x02 - it should be answered by the message for company B I've noticed that in kern.log the incoming MSN is logged, but didn't find anything in the * log nor in the *CLI screen logs Is there a way to manage this in asterisk in order to make the different welcomes using only different contexts (let say, [main-menu-A] and [main-menu-B] )? Tnx, -- Stefano Finetti Technical Coordinator Lynx Automotive srl [EMAIL PROTECTED] Tel: 199 79 79 30 Fax: 06 233 227 934 Linux Registered User #271978 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming CallerID management
Stefano Finetti wrote: Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a different welcome message. If a customer calls the 06x01 - it should be answered by the message for company A If a customer calls the 06x02 - it should be answered by the message for company B I've noticed that in kern.log the incoming MSN is logged, but didn't find anything in the * log nor in the *CLI screen logs Is there a way to manage this in asterisk in order to make the different welcomes using only different contexts (let say, [main-menu-A] and [main-menu-B] )? Tnx, what incomming channel do you use, how is your * connected to the world ? -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming CallerID management
I do the same thing..
I use chan_capi and the MSN numbers on my ISDN line..
Firstly all inbound calls are placed into my [default] context and then redirected
from there to the company specific start context.. Then you setup your company
specific intro message from there..
Example
[default]
; Inbound call redirection based on MSN
exten = {MSN NumberA},1,Goto(companyA-start,s,1)
exten = {MSN NumberB},1,Goto(companyB-start,s,1)
[companyA-start]
;Context for CompanyA
exten = s,1,...
[companyB-start]
;Context for CompanyB
exten = s,1,...
There may very well be an easier way to do this but this worked fine for me..
Later..
Greetings,
I need if possibile an explanation on how to manage the incoming callerid
for an incoming call. Let me explain the situation:
We have two different companies in this office that shares the same PBX (*
box). Each company have its own number for the incoming calls.
What i'd like to implement is something that, depending on the incoming line
that is involved in the call, plays a different welcome message.
If a customer calls the 06x01 - it should be answered by the message
for company A
If a customer calls the 06x02 - it should be answered by the message
for company B
I've noticed that in kern.log the incoming MSN is logged, but didn't find
anything in the * log nor in the *CLI screen logs
Is there a way to manage this in asterisk in order to make the different
welcomes using only different contexts (let say, [main-menu-A] and
[main-menu-B] )?
Tnx,
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 06 233 227 934
Linux Registered User #271978
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[Asterisk-Users] Help configuring E400P cards
Title: Carlos Fernández Puente Hi everybody. We have a problem with the configuration of the card, the cards work and we receive incoming calls but asterisk don't receive dnid. We have 5 servers with 1 E400P with the same problem and the telco told us that we need to configure the card to request it, how we can do this? Can you help me to solve the problem. Best regards, Carlos Fernández Puente [EMAIL PROTECTED] Ingeniero de proyectosAlisysSoftware Alisys Software, S.L. Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 - 915678474 Fax: 915714244 web: http://www.alisys.net wap: http://www.alisys.net/wap/
Re: [Asterisk-Users] Incoming CallerID management
From: Pavel Litvinenko [EMAIL PROTECTED] what incomming channel do you use, how is your * connected to the world ? Ooops! I just forgot to write that part of the email :-) My * box is connected with 4 passive ISDN HFC cards (hisax driver). So, I've 8 ttyIs usable. When i receive a Call, i see only the ttyI device that is ringing, not the number. -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
Strange.. I had a symbolic link, and it wouldn't work. After I finally got it working properly, i even tried to remove it from /usr/bin and symlink it, and it wouldn't work again... couldn't for the life of me figure out why. - Original Message - From: Joseph Finley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 4:20 PM Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remotely picked-up extension keeps ringing
Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, -- There are no Debian developers in any part of Hell, because the good karma incurred by being one takes you straight to the pearly gates. Of course, the frequent flame wars you put up with on the Debian lists make up for this on Earth. - Seth Cohn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't use 2 controllers
Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. -- ___ Simone Vasoli BK s.r.l. - Brain and Knowledge e-mail: simone.vasoli[at]b-k.it cell: +39 348 0830539 tel: 0187 1874200 ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P in Switzerland
Hi all ! Have some of you some experience with Basilisk, Wildcard E100P and Swisscom (Swizterland) ? Are all these working well together ? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over SIP
On Thu, 2003-09-04 at 01:18, Ing. Angel Gomez Garcia wrote: Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line - mp108fxo - Asterisk -- mp108fxs --- fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid extension ff in context ). 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call to this extension but when the fax answers the call is dropped ( don't have here the SIP debug output ) but seems that when * tries to make the bridge the mp108 fxo sends a BYE. 3) I dialed in to * with a phone ( external line and internal extension ) and dial extension for the fax and i cann hear the fax, so the call is not dropped, the bridge is established successfully. 4) If I pickup the call on the fax machine ( it has a phone set ) and then pressed the 'start' button to start de fax receiver, then, the two faxes talked to each other and the fax is received well. By listening to #4 it sounds like you answered your own question, yes it is possible. Now you need to find out why in #2 that it sends a bye. My guess is that you have a difference in the dial command options that keep asterisk listening to the line when dialing the extension that isn't there on the ff extension. This may have asterisk trying to issue a reinvite to connect the call legs together without asterisk in the middle. This is causing the BYE, and then everything fall apart. Maybe you need to make sure the canreinvite is turned off for this device in the sip.conf and try some more. Seems that the problem is only when the fax answer automatically ( could be the tones the receiving fax plays ? ), the same problem happens when i try to use hylafax to receive the fax. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't use 2 controllers
Simone Vasoli (BK s.r.l.) wrote: Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. You can only use one Fritz PCI card per box due to a limitation built into the AVM CAPI driver. However you can get around this by hacking the (binary) fcpci kernel module and changing the name of second and subsequent modules. I haven't tried it personally, but I think there are several people who are using this technique sucessfully :). Just in case anyone is interested, AVM also say that you can't use the Fritz card and the B1 card in the same box. However I have found it seems to work fine provided the B1 CAPI driver is loaded *after* the Fritz driver. -- Jamie Neil | [EMAIL PROTECTED] | 0870 454 Versado I.T. Services Ltd. | http://versado.net/ | 0845 450 1254 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
-Original Message- From: Joseph Finley [mailto:[EMAIL PROTECTED] Sent: 3. september 2003 23:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh Ok I get same results when using Answer, so I'll just stick with that thx Mickey - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't use 2 controllers
I did as you tell me, infact both controllers works (I receive incoming calls in both channels). The problem is when I want to make 3 calls, the first and the second are controlled by the 2 controller, but the third too (but obviously all the channels are busy). How can I say to asterisk to use my first controller? Thank you... Il gio, 2003-09-04 alle 11:40, Jamie Neil ha scritto: Simone Vasoli (BK s.r.l.) wrote: Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. You can only use one Fritz PCI card per box due to a limitation built into the AVM CAPI driver. However you can get around this by hacking the (binary) fcpci kernel module and changing the name of second and subsequent modules. I haven't tried it personally, but I think there are several people who are using this technique sucessfully :). Just in case anyone is interested, AVM also say that you can't use the Fritz card and the B1 card in the same box. However I have found it seems to work fine provided the B1 CAPI driver is loaded *after* the Fritz driver. -- ___ Simone Vasoli BK s.r.l. - Brain and Knowledge e-mail: [EMAIL PROTECTED] cell: +39 348 0830539 tel: 0187 1874200 ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't use 2 controllers
Da t, 2003-09-04 at 11:40, Jamie Neil napsal: Simone Vasoli (BK s.r.l.) wrote: Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. You can only use one Fritz PCI card per box due to a limitation built into the AVM CAPI driver. check this link - is describe how to use more than one fcpci in pc. I personally use 3 cards im my computers, without problems... http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO However you can get around this by hacking the (binary) fcpci kernel module and changing the name of second and subsequent modules. I haven't tried it personally, but I think there are several people who are using this technique sucessfully :). Just in case anyone is interested, AVM also say that you can't use the Fritz card and the B1 card in the same box. However I have found it seems to work fine provided the B1 CAPI driver is loaded *after* the Fritz driver. -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - DTMF Payload type
I have a problem with my Welltech Wellgates. I can't call any extension which starts with or includes * or #. When dialing it responds fine but after some seconds I just get a busy tone and on the Asterisk console it says SIP/2.0 484 Address Incomplete. Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received I then tried to set it to 101 (found this value somewhere on the net) and verified that voice responds now worked, but I don't know if this is the correct type? Still I can't use * or # regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PCI master Abort
i had a T100P working perfectly when on my asterisk box. but to meet our current requirements i added another T100P to set up another channel bank to use with it. after a couple of minutes of operation i get the multiple PCI Master Abort messages on the screen and asterisk freezes. is there any way out this state? please i will know if anyone has a solution to this problem. My email is :- [EMAIL PROTECTED] -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP - DTMF Payload type
Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received Just wanted to note I just observed it doesn't send any number at all when using # or *. In the field Contact it writes: sip:@10.1.1.51 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
Along the same lines, though, I do agree that MP3Player app should cause the system to trigger the answer without having to do it manually, but I could see where you might not want it to, as well. In theory, the Playback app (pardon, this is what i've gathered by toying with it) triggers the 'answer' function if the call is not already answered. Couldn't we get the MP3Player app to do the same? I'm not that skilled of a programmer, otherwise, I'd hack it up and do it myself. -Josh - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 4:34 AM Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer -Original Message- From: Joseph Finley [mailto:[EMAIL PROTECTED] Sent: 3. september 2003 23:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh Ok I get same results when using Answer, so I'll just stick with that thx Mickey - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ 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 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] Arraycom voip phone
Hi All, Does anyone have any experience with the ArrayCom VoIP phone? I bought one a couple of weeks ago, it used to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a master reset? PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't use 2 controllers
I checked the same link (http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO), and the two fritzcards work when I receive calls, but when I make a call (or more calls), asterisk only uses my second controller Il gio, 2003-09-04 alle 11:53, Marian Danisek ha scritto: Da t, 2003-09-04 at 11:40, Jamie Neil napsal: Simone Vasoli (BK s.r.l.) wrote: Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. You can only use one Fritz PCI card per box due to a limitation built into the AVM CAPI driver. check this link - is describe how to use more than one fcpci in pc. I personally use 3 cards im my computers, without problems... http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO However you can get around this by hacking the (binary) fcpci kernel module and changing the name of second and subsequent modules. I haven't tried it personally, but I think there are several people who are using this technique sucessfully :). Just in case anyone is interested, AVM also say that you can't use the Fritz card and the B1 card in the same box. However I have found it seems to work fine provided the B1 CAPI driver is loaded *after* the Fritz driver. -- ___ Simone Vasoli BK s.r.l. - Brain and Knowledge e-mail: [EMAIL PROTECTED] cell: +39 348 0830539 tel: 0187 1874200 ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI master Abort
Hi, I have had the same problem. Turned out to be that my digium card was sharing irq with a different card allready in the machine. If you give your T100P it's own dedicated IP, i bet the problem will be solved. Greetings, Tj - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:28 PM Subject: [Asterisk-Users] PCI master Abort i had a T100P working perfectly when on my asterisk box. but to meet our current requirements i added another T100P to set up another channel bank to use with it. after a couple of minutes of operation i get the multiple PCI Master Abort messages on the screen and asterisk freezes. is there any way out this state? please i will know if anyone has a solution to this problem. My email is :- [EMAIL PROTECTED] -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ 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] PCI master Abort
Sorry offcourse i meant IRQ not IP :) - Original Message - From: Tjardick van der Kraan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:40 PM Subject: Re: [Asterisk-Users] PCI master Abort Hi, I have had the same problem. Turned out to be that my digium card was sharing irq with a different card allready in the machine. If you give your T100P it's own dedicated IP, i bet the problem will be solved. Greetings, Tj - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:28 PM Subject: [Asterisk-Users] PCI master Abort i had a T100P working perfectly when on my asterisk box. but to meet our current requirements i added another T100P to set up another channel bank to use with it. after a couple of minutes of operation i get the multiple PCI Master Abort messages on the screen and asterisk freezes. is there any way out this state? please i will know if anyone has a solution to this problem. My email is :- [EMAIL PROTECTED] -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ 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] oh323 - sip communication problem
I've got problem with connections h323 - sip and sip - h323. I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5 When I call from Cisco (SIP) to h323 node by alias registered on gatekeeper and h323 node will answer the phone... I have on my Cisco still Ringing. Call termination, no matter from which side works fine. koga*CLI oh323 show info koga*CLI ^^^ why is here empty line? :) Information about active OpenH323 channel(s) Num. Token State Init RX/TX Format Remote RTP Addr. Local RTP Addr. 0 ip$localhost/21538 RINGLocal 0/160 NULL 0.0.0.0:0 0.0.0.0:0 koga*CLI show channels Channel (ContextExtensionPri ) State Appl. Data H323:21538 (voip-h323 s1 ) Ringing AppDial (Outgoing Line) SIP/blue-ebfa (defaultmarosin 2 )Ring Dial OH323/marosin I can't call from h323 phone to sip. I get that user is not registered on gatekeeper... My configuration: oh323.conf: [register] alias=asterisk alias=123 alias=blue alias=blues ;alias=marosin extensions.conf: [voip-h323] exten = marosin,1,Ringing exten = marosin,2,Dial,OH323/marosin exten = marosin,3,Hangup marosin is h323 phone blues and blue are sip phones. -- pozdr. Pawe Goaszewski - worth to see: http://www.againsttcpa.com/ CPU not found - software emulation... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help configuring E400P cards
Carlos Fernández Puente wrote:
We have a problem with the configuration of the card, the cards work
and we receive incoming calls but asterisk don't receive dnid. We
have 5 servers with 1 E400P with the same problem and the telco told
us that we need to configure the card to request it, how we can do
this?
Hi Carlos. When you say don't receive dnid, what exactly do you mean?
If you start Asterisk with -vgc, and make a call into the box,
does it give you any information?
For example, on my E400P, I get:
Connected to Asterisk CVS-08 currently running on vampire (pid = 23767)
-- Accepting call from 'x' to 'y' on channel 31, span 1
(Where x and y have been changed from the actual numbers,
obviously, with y being the DNID.)
Or are you referring to the ${DNID} variable not being set?
I've patched my installation to set ${DNID}. I'm currently using it to
route calls via an external AGI database lookup. I will submit a patch
to the dev-list later today for consideration for CVS if you like.
--
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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[Asterisk-Users] Call script after hangup
Beginner: How can a script be called after a
calling user hangup?
What's wrong with this:
[incoming]
exten = s,1,Playback,welcome
exten = s,2,Record,msgfile:gsm
exten = h,1,Goto(callscript,1,1)
[callscript]
exten = 1,1,Wait,5
exten = 1,2,System("SomeScript")
Thank you
AW: [Asterisk-Users] Call script after hangup
Hi
Frank,
why
you so complicated ?
Try
following:
[incoming]
exten =
s,1,Playback,welcome
exten =
s,2,Record,msgfile:gsm
exten = h,1,System(/home/frank/callscript.pl)
as
sample ... :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-Von:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Im Auftrag von Frank
N.Gesendet: Donnerstag, 4. September 2003 14:54An:
[EMAIL PROTECTED]Betreff: [Asterisk-Users] Call
script after hangup
Beginner: How can a script be called after a
calling user hangup?
What's wrong with this:
[incoming]
exten = s,1,Playback,welcome
exten = s,2,Record,msgfile:gsm
exten = h,1,Goto(callscript,1,1)
[callscript]
exten = 1,1,Wait,5
exten = 1,2,System("SomeScript")
Thank you
Re: [Asterisk-Users] IAX2 ports usage
RTP ports are not applying to IAX/IAX2. Martin On Thu, 4 Sep 2003, WipeOut . wrote: Yes, The RTP ports in * are configurable in rtp.conf.. The default is 1 - 2 Later HI! but when making iax2 calls, a packet monitor would only reveal this UDP port. (Between two * servers) ?? 4569 proto: U ( I would assume even the RTP headers get enclosed by UDP, so there should have been more UDP port variants. Not the case when monitored.) I've got these in my rtp.conf rtpstart=1 rtpend=2 Does it mean RTP use the above udp port range ?( 1~2). denzel - Original Message - From: Wade J. Weppler To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 8:42 AM Subject: RE: [Asterisk-Users] IAX2 ports usage The RDP packets need to be dealt with as well. They are specified in rtp.conf -wade -Original Message- From: denzel-infotechs [mailto:[EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX2 ports usage hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2. DenZel. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Help configuring E400P cards
Turn off immediate=yes in zapata.conf regards Martin On Thu, 4 Sep 2003, =?us-ascii?Q?Carlos Fern=E1ndez Puente?= wrote: Hi everybody. We have a problem with the configuration of the card, the cards work and we receive incoming calls but asterisk don't receive dnid. We have 5 servers with 1 E400P with the same problem and the telco told us that we need to configure the card to request it, how we can do this? Can you help me to solve the problem. Best regards, Carlos Fernández Puente [EMAIL PROTECTED] Ingeniero de proyectos Alisys Software Alisys Software, S.L. Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 - 915678474 Fax: 915714244 web: http://www.alisys.net wap: http://www.alisys.net/wap/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call script after hangup
Thomas,
your
suggestion does work better.
However, I doesn't solve my problem. Here is
callscript.pl:
#!/usr/bin/perlprint "waiting...\n";sleep
5;`cp /usr/src/asterisk/sample.call /var/spool/asterisk/outgoing`;print
"call created\n";
The
problem is the incoming and outgoing calls are made on the same channel
(Zap/1).
I
believe the porblem is that, since the incoming call is not "closed"
before the outgoing call is created, the outgoing call does not
work.
I was
hoping the delay would solve this problem... but obviously it
doesn't.
Do you
have any suggestions?
Thank
you.
-Original Message-From: Thomas Haeger
[mailto:[EMAIL PROTECTED] Sent: 4 septembre, 2003
09:42To: [EMAIL PROTECTED]Subject: AW:
[Asterisk-Users] Call script after hangup
Hi
Frank,
why
you so complicated ?
Try
following:
[incoming]
exten =
s,1,Playback,welcome
exten =
s,2,Record,msgfile:gsm
exten = h,1,System(/home/frank/callscript.pl)
as
sample ... :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-Von:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Im Auftrag von Frank
N.Gesendet: Donnerstag, 4. September 2003 14:54An:
[EMAIL PROTECTED]Betreff: [Asterisk-Users] Call
script after hangup
Beginner: How can a script be called after a
calling user hangup?
What's wrong with this:
[incoming]
exten = s,1,Playback,welcome
exten = s,2,Record,msgfile:gsm
exten =
h,1,Goto(callscript,1,1)
[callscript]
exten = 1,1,Wait,5
exten =
1,2,System("SomeScript")
Thank
you
RE: [Asterisk-Users] Call script after hangup
Hey
guys,
I've
been trying to do something similar. Basically I want to call a script with AGI
every time a call is hungup, regardless of who hangs up. The purpose of which is
to record call end time and duration in another app.
is 'h'
a reserved extension number for capturing hangups?
TIA,
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Thomas
HaegerSent: 04 September 2003 14:42To:
[EMAIL PROTECTED]Subject: AW: [Asterisk-Users] Call
script after hangup
Hi
Frank,
why
you so complicated ?
Try
following:
[incoming]
exten =
s,1,Playback,welcome
exten =
s,2,Record,msgfile:gsm
exten = h,1,System(/home/frank/callscript.pl)
as
sample ... :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-Von:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Im Auftrag von Frank
N.Gesendet: Donnerstag, 4. September 2003 14:54An:
[EMAIL PROTECTED]Betreff: [Asterisk-Users] Call
script after hangup
Beginner: How can a script be called after a
calling user hangup?
What's wrong with this:
[incoming]
exten = s,1,Playback,welcome
exten = s,2,Record,msgfile:gsm
exten =
h,1,Goto(callscript,1,1)
[callscript]
exten = 1,1,Wait,5
exten =
1,2,System("SomeScript")
Thank
you
RE: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0?
Rich, you can do **# and go into the Network config and hit **# again, you should notice the LockPad come unlocked and then you can make changes. If you upgraded, the default password is cisco Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, September 03, 2003 9:58 PM To: Asterisk-users-list Subject: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0? Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't unlock it to do any configs. Any thoughts? Rich [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
[Asterisk-Users] Asterisk vs. Vocal (Vovida) vs. Bayonne
Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a vi guy with CLI preferences. Gastman handles monitoring (astman too) but it doesn't handle the configuration piece (that I see?) I've seen some rumors/proof of concepts to PHP web-based front end to configuration? (Basically create and extension, voicemail box, etc.. -- I know it would be fairly trivial, but... Has anyone tackled that?) Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about cdr_sql fields
Hello- Is it possible to set the CDR record field called accountcode from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
For the benefit of others having this problem - I installed the latested CVS build and the problem went away - I can hear audio now from X-lite. I was using the debian unstable package. Here's what I did: cd /usr/src mkdir asterisk export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (supply the anoncvs password here when prompted) cvs checkout asterisk cd asterisk make make install make samples make sounds (I think that's right - memory's getting fuzzy from age) added my extention back into sip.conf works like a champ. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about cdr_sql fields
Sure is, you can set the accountcode=13213 within each entity of sip.conf (or iax.conf I believe). -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: 04 September 2003 17:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about cdr_sql fields Hello- Is it possible to set the CDR record field called accountcode from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call script after hangup
Frank N. wrote: I believe the porblem is that, since the incoming call is not closed before the outgoing call is created, the outgoing call does not work. I was hoping the delay would solve this problem... but obviously it doesn't. No - it still won't relinquish the call until the hangup handler has completed. What you need to do is to have the AGI script return, such that the call exits. Then five seconds later, copy the file. You could do this by setting up a BASH script which executed the Perl in the background. I.e. #!/bin/sh /path/to/script/foo.pl Make sense? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First steps with asterisk and Voip (SIP, MGCP, H323, IAX).
Hi everybody, Now I want to began use Asterisk in orden to have VoIP and to use at gateway between the PSTN network and IP phones. Any one have some link to have more documentation about. Thks a lot.
Re: [Asterisk-Users] RedHat Distribution
Redhat 9 works fine unless you really need G729 working on H323 in which case the only solution seems to be chanh323, which only works with G729 support on Redhat 8 .. I found out the hard way :) cheers Dave - Original Message - From: Ernest W. Lessenger To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 5:45 PM Subject: Re: [Asterisk-Users] RedHat Distribution At 12:29 PM 9/2/2003 +0100, you wrote: I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend.Redhat 9 works perfectly. Install with the kernel sources and devel libraries, and the developers software, i.e. gcc, and upgrade to most recent rpms before making asterisk.--Ernest
Re: [Asterisk-Users] IAX2 ports usage
Denzel said: HI! but when making iax2 calls, a packet monitor would only reveal this UDP port. (Between two * servers) ?? 4569 proto: U This is correct. IAX uses 5036 and IAX2 uses 4569, ONLY. Both include signalling *and* voice, there is no RTP with these transports. ( I would assume even the RTP headers get enclosed by UDP, so there should have been more UDP port variants. Not the case when monitored.) RTP is UDP based protocol, it uses dynamically assigned UDP ports for communication. IAX/IAX2 do NOT use RTP. I've got these in my rtp.conf rtpstart=1 rtpend=2 Does it mean RTP use the above udp port range ?( 1~2). Yes, for all protocols that use RTP as a transport (SIP, H.323, etc). From: Wade J. Weppler The RDP packets need to be dealt with as well. No RTP packets with IAX/IAX2. hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2. The correct answer to your question the first time should have been No, there are no other ports to worry about. Someone please correct me if I'm wrong. -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Scalability of the Asterisk
Hello, Is there any limit on number of clients (Analog/IP Phones) asterisk can serve ?. Can it scale to 5000 clients or more, any real world statistics will be of great use. Thanks, Tarun ___ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://rediff.com/matchmaker?2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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] Asterisk vs. Vocal (Vovida) vs. Bayonne
On Thu, 2003-09-04 at 09:43, [EMAIL PROTECTED] wrote: Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. search the archive with google. It is there already. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a vi guy with CLI preferences. Gastman handles monitoring (astman too) but it doesn't handle the configuration piece (that I see?) PBX configuration is not for a point and drool idiot. You must make intelligent considerations for call volume, deployment of equipment to satisfy needs, and understanding of how adding extensions in one area changes what is available in others unless it is segregated appropriately. I've seen some rumors/proof of concepts to PHP web-based front end to configuration? (Basically create and extension, voicemail box, etc.. -- I know it would be fairly trivial, but... Has anyone tackled that?) It is possible to do it, it is possible to make a generic extension logic that allows extensions to be made simply by creating a voice mailbox. But see above comment on why that isn't really a good thing to do. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
Parking the call is a problem becasue you will not hear the parked call location (because its a blind transfer into the parked call).. The only solution I could think of is to call the person you want to transfer to on the second line, then go back to the first line and blind transfer the call.. (the person you are transfering to will have to hang up after you have spoken to them) What is the process for transfering with the flash button?? I have always used the transfer button and the redial/send button.. no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about cdr_sql fields
Hi, astro*CLI show application SetAccount astro*CLI -= Info about application 'SetAccount' =- [Synopsis]: Sets user language [Description]: SetAccount([account]): Set the channel account code for billing purposes. Always returns 0. astro*CLI Although there's a little bug in the Synopsis ;) matteo. Il gio, 2003-09-04 alle 17:10, Scott Stingel ha scritto: Hello- Is it possible to set the CDR record field called accountcode from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call script after hangup
a stupid idea is... why don't lauch the script in the background ? so asterisk could close the chan (since lauching it into the bg * gets the control immediately back) . just my 2 cents. Matteo. Il gio, 2003-09-04 alle 16:29, Frank N. ha scritto: Thomas, your suggestion does work better. However, I doesn't solve my problem. Here is callscript.pl: #!/usr/bin/perl print waiting...\n; sleep 5; `cp /usr/src/asterisk/sample.call /var/spool/asterisk/outgoing`; print call created\n; The problem is the incoming and outgoing calls are made on the same channel (Zap/1). I believe the porblem is that, since the incoming call is not closed before the outgoing call is created, the outgoing call does not work. I was hoping the delay would solve this problem... but obviously it doesn't. Do you have any suggestions? Thank you. -Original Message- From: Thomas Haeger [mailto:[EMAIL PROTECTED] Sent: 4 septembre, 2003 09:42 To: [EMAIL PROTECTED] Subject: AW: [Asterisk-Users] Call script after hangup Hi Frank, why you so complicated ? Try following: [incoming] exten = s,1,Playback,welcome exten = s,2,Record,msgfile:gsm exten = h,1,System(/home/frank/callscript.pl) as sample ... :-) Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Frank N. Gesendet: Donnerstag, 4. September 2003 14:54 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Call script after hangup Beginner: How can a script be called after a calling user hangup? What's wrong with this: [incoming] exten = s,1,Playback,welcome exten = s,2,Record,msgfile:gsm exten = h,1,Goto(callscript,1,1) [callscript] exten = 1,1,Wait,5 exten = 1,2,System(SomeScript) Thank you -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
Dave, Would you describe how to acheive that? I can't have this configuration working. Daniel Dave Alan Caruana a crit: no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: "Daniel ANDRE" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a crit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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 -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
I was referring to the transfer button (sorry) what again is the way you are using to transfer calls ? so far what i'm doing is after accepting a call, parking it .. then phoning the guy who wants the call and telling him the call is parked on 701 for example .. cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 7:08 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Parking the call is a problem becasue you will not hear the parked call location (because its a blind transfer into the parked call).. The only solution I could think of is to call the person you want to transfer to on the second line, then go back to the first line and blind transfer the call.. (the person you are transfering to will have to hang up after you have spoken to them) What is the process for transfering with the flash button?? I have always used the transfer button and the redial/send button.. no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Scalability of the Asterisk
Hello, Is there any limit on number of clients (Analog/IP Phones) asterisk can serve ?. Can it scale to 5000 clients or more, any real world statistics will be of great use. Thanks, Tarun I don't think there is any limit on the number of extensions it would be more to do with the number of analog ports you could connect to the system or the number of ethernet ports you could get on your network.. The more important question would be to ask how many concurrent calls the system would handle at any time (also dependent on codecs used).. If you have 5000 clients but no one is making a call then your system could be quite small.. If you dig through the mailing list you will se that hardware questions have been asked many many times.. maybe you could get some useful info from those replies.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
has anyone got G729 and SIP working together? some config examples would help :) since I need to do this at a client where I don't really have internet access, or the will to root around mailing lists with the client breathing down my neck! thsnk Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 7:13 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Call script after hangup
is to say... I don't know if you can write a sort of exten = h,1,System(/home/frank/callscript.pl ) but surely you can do exten = h,1,System(/home/frank/callscript.sh) where callscript.sh is #!/bin/sh /home/frank/callscript.pl Probably (and surely I think) there's a better method to fork into the bg directly from perl... but I'm not a perl guru ;( matteo. Il gio, 2003-09-04 alle 19:17, Brancaleoni Matteo ha scritto: a stupid idea is... why don't lauch the script in the background ? so asterisk could close the chan (since lauching it into the bg * gets the control immediately back) . just my 2 cents. Matteo. Il gio, 2003-09-04 alle 16:29, Frank N. ha scritto: Thomas, your suggestion does work better. However, I doesn't solve my problem. Here is callscript.pl: #!/usr/bin/perl print waiting...\n; sleep 5; `cp /usr/src/asterisk/sample.call /var/spool/asterisk/outgoing`; print call created\n; The problem is the incoming and outgoing calls are made on the same channel (Zap/1). I believe the porblem is that, since the incoming call is not closed before the outgoing call is created, the outgoing call does not work. I was hoping the delay would solve this problem... but obviously it doesn't. Do you have any suggestions? Thank you. -Original Message- From: Thomas Haeger [mailto:[EMAIL PROTECTED] Sent: 4 septembre, 2003 09:42 To: [EMAIL PROTECTED] Subject: AW: [Asterisk-Users] Call script after hangup Hi Frank, why you so complicated ? Try following: [incoming] exten = s,1,Playback,welcome exten = s,2,Record,msgfile:gsm exten = h,1,System(/home/frank/callscript.pl) as sample ... :-) Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Frank N. Gesendet: Donnerstag, 4. September 2003 14:54 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Call script after hangup Beginner: How can a script be called after a calling user hangup? What's wrong with this: [incoming] exten = s,1,Playback,welcome exten = s,2,Record,msgfile:gsm exten = h,1,Goto(callscript,1,1) [callscript] exten = 1,1,Wait,5 exten = 1,2,System(SomeScript) Thank you -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Stops Responding
1) 8/29 2) The Agents are not used to * to hang up. I think it works until they hangup the phone, and then when they pick up again to login there is nothing (Dialtone, dtmf,...) 3) Can't hard hang up, see #2 :) 4) masqueraded? Not sure what you mean here. We use only Zap devices. No, VOIP. 5) nope. From my paste before. 25 and 26 are inbound/remote calls. 52, 54, 64, 65, 66 are all agents/local. The local phones would not work. But not ALL of them. It has only happened on agent phones. Everyone else in the building seems to work fine. Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) On Wednesday, September 3, 2003, at 10:27 AM, TC wrote: John 1) Is this from current CVS ??? 2) does the agent notice by the fact that they can't do a * to hang up the channel, in fact all dtmf is not recognized 3) if you do a hard hang up the agent line does it stay up 4) Does it only happend when the call is masqueraded to the agent line 5) if the remote hangs up the channel does the agent line come free If you ans yes to these items, I beleive I have duplicated this in testing last night with a config I was testing last night ... or is this a system wide deadlock ?? Can you do any other * functions outside of queues and agents, like dial an extension etc -Original Message- From: John Congdon [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: September 3, 2003 5:43 AM Subject: [Asterisk-Users] Asterisk Stops Responding This is getting to be a big problem. I am hoping it is something I have setup wrong somewhere... Various channels just freeze. It always appears to be the agents phones only. They will come to me and say the phones are down again. This morning here is what I see. I can not do STOP NOW. Just returns to the CLI prompt. I have to kill it. Notice that I try to hangup the channels and nothing happens. Any suggestions? ___ 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] IVR only system with scalibility with asterisk???
Hello all: Thank you for taking the time to read this post. Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from http://search.cpan.org/author/MUKUND/. We are having limited success. I found the asterisk software and have trugded through the last several months looking for IVR specific comments with minimal success. Issues: 1. We need to have a working system by yesterday (since we were told yesterday ;) my problem not yours). Realy, how easy is asterisk to develop for in a IVR message - response - authorize/validate - contiune scenario? We will need to do database lookups. 2. We expect that we will end up greater than 100 users that will call in 3 or 4 time each day during (approx) normal business hours in the next couple of months. We also have the possibility that the next step may involve several hundred users. How can I provide something now and scale UP from a commidity PC (running GNU/Linux of course)? The Wildcard X100P only has 1 port. Are there other higher density options that just work? I've seen mentioned an Intel/Dialogic card that looks high density and expensive and interesting. I don't mind having a farm of these things on commidity hardware... within reason. Again, I'm a newbe trying to get myself up to speed on this topic. 3. I need to provide a working model very soon. What is cheapest way to put together a system with AVAILIBLE parts? There seems to be a shortage of some of the cards on the yahoo store front. So, looking for everything... Thank you very much. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
You should be able to simply use either allow=all in sip.conf and then change the codec order on the phone.. or if you want a little more control you can put disallow=all allow=g729 allow=... in the sip.conf and selectivly allow the codecs.. you will probably still want to set the order on the phone as well.. You may need to make sure on the allow=g729 syntax.. I typed this from memory so it could be wrong.. has anyone got G729 and SIP working together? some config examples would help :) since I need to do this at a client where I don't really have internet access, or the will to root around mailing lists with the client breathing down my neck! thsnk Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 7:13 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR only system with scalibility with asterisk???
On Thu, 4 Sep 2003, PJ Welsh wrote: Thank you for taking the time to read this post. you're very welcome, welcome to the community, its extremely helpful Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from http://search.cpan.org/author/MUKUND/. We are having limited success. I found the asterisk software and have trugded through the last several months looking for IVR specific comments with minimal success. you've found a right solution... Issues: 1. We need to have a working system by yesterday (since we were told yesterday ;) my problem not yours). Realy, how easy is asterisk to develop for in a IVR message - response - authorize/validate - contiune scenario? We will need to do database lookups. i'd go as far as to say that developing IVR through * is a dream. simple to a fault almost at times... db integration, hell, any type of integration is as simple as it can get, with a little help (get on #asterisk) you'll have your IVR up yesterday 2. We expect that we will end up greater than 100 users that will call in 3 or 4 time each day during (approx) normal business hours in the next couple of months. We also have the possibility that the next step may involve several hundred users. How can I provide something now and scale UP from a commidity PC (running GNU/Linux of course)? The Wildcard X100P only has 1 port. Are there other higher density options that just work? I've seen mentioned an Intel/Dialogic card that looks high density and expensive and interesting. I don't mind having a farm of these things on commidity hardware... within reason. Again, I'm a newbe trying to get myself up to speed on this topic. digium makes single and quad port E/T1 cards too, very economical, and scale wonderfully well.. 3. I need to provide a working model very soon. What is cheapest way to put together a system with AVAILIBLE parts? There seems to be a shortage of some of the cards on the yahoo store front. depends on the number of lines, you can setup all the IVR functionality and test it though VoIP while youre waiting for the cards, and as soon as they arrive, go live on tdm circuits as well... -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR only system with scalibility with asterisk???
Issues: 1. We need to have a working system by yesterday (since we were told yesterday ;) my problem not yours). Realy, how easy is asterisk to develop for in a IVR message - response - authorize/validate - contiune scenario? We will need to do database lookups. An AGI application should be able to do this for you.. 2. We expect that we will end up greater than 100 users that will call in 3 or 4 time each day during (approx) normal business hours in the next couple of months. We also have the possibility that the next step may involve several hundred users. How can I provide something now and scale UP from a commidity PC (running GNU/Linux of course)? The Wildcard X100P only has 1 port. Are there other higher density options that just work? I've seen mentioned an Intel/Dialogic card that looks high density and expensive and interesting. I don't mind having a farm of these things on commidity hardware... within reason. Again, I'm a newbe trying to get myself up to speed on this topic. The digium hardware is your best bet (as opposed to intel/dialogic), you can get a single quad port card that will handle either 96 channels of T1 or 120 channels of E1 depending on which part of the world you live in.. 3. I need to provide a working model very soon. What is cheapest way to put together a system with AVAILIBLE parts? There seems to be a shortage of some of the cards on the yahoo store front. An X100P will give you a working model with a simgle channel, If you want to use a BRI ISDN line you could use an AVM or EICON ISDN card.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR only system with scalibility with asterisk???
i'd go as far as to say that developing IVR through * is a dream. simple to a fault almost at times... db integration, hell, any type of integration is as simple as it can get, with a little help (get on #asterisk) you'll have your IVR up yesterday That's really good to hear - my background is in IVR and I've been developing with Intel's CT-ADE (formerly VOS from Parity Software) for quite a while now. I'm starting to play with Asterisk and liking everything I see so far, so to hear you say that is really encouraging. One thing I do find a bit lacking is documentation, but being a developer I know how we all hate writing docs ;-) I've seen the AGI HTML dump, think it makes sense, I guess I just need to do a bit more playing and looking at the examples. digium makes single and quad port E/T1 cards too, very economical, and scale wonderfully well.. Yes, I love the pricing compared to Dialogic cards! When you say they scale well, how many cards are we talking about in a single chassis? Regards Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR only system with scalibility with asterisk???
On Thu, 2003-09-04 at 12:43, PJ Welsh wrote: Hello all: Thank you for taking the time to read this post. Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from http://search.cpan.org/author/MUKUND/. We are having limited success. I found the asterisk software and have trugded through the last several months looking for IVR specific comments with minimal success. Issues: 1. We need to have a working system by yesterday (since we were told yesterday ;) my problem not yours). Realy, how easy is asterisk to develop for in a IVR message - response - authorize/validate - contiune scenario? We will need to do database lookups. 2. We expect that we will end up greater than 100 users that will call in 3 or 4 time each day during (approx) normal business hours in the next couple of months. We also have the possibility that the next step may involve several hundred users. How can I provide something now and scale UP from a commidity PC (running GNU/Linux of course)? The Wildcard X100P only has 1 port. Are there other higher density options that just work? I've seen mentioned an Intel/Dialogic card that looks high density and expensive and interesting. I don't mind having a farm of these things on commidity hardware... within reason. Again, I'm a newbe trying to get myself up to speed on this topic. 3. I need to provide a working model very soon. What is cheapest way to put together a system with AVAILIBLE parts? There seems to be a shortage of some of the cards on the yahoo store front. First you need to decide on how many ports you will need, how important ease of scalability is. For the number of ports, you need to decide how much tolerance you have for the people remotely to deal with a busy signal. So far you mentioned 45 people making 3-4 calls a day over a ~8 hour day. The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and everyone waited till it became available. My guess is you don't want people on redial that often and waiting for the port to come open. Next, you move on to what is the acceptable idle amount of service available. If you scaled up to say 5 lines, and the call length is short, then you will have your service mostly idle, but it can handle peak times better. I'll let you continue this line of questioning internally. Next to decide on hardware, if you think you may need more than 10 lines, you need to move to digital trunks. You can start with a T100P and a channel bank until your costs justify switching over to a T1. The benefit is already having the hardware in hand and used to it while on spending a little more short term to get the FXO channel bank that you will either sell off later, or convert to FXS for internal extensions if you want to switch services. If you already have a PBX in house and can drop a T1 interface to you asterisk box, that is good too. As for your application. You mention looking into perl modules, so I assume you have some perl familiarity. From AGI you can script up any database access and prompting you so wish to undertake. Essentially it will come out to be something like. stream file(prompt) while (not enough digits) wait for digits collect dialed digits validate(digits) # in this sub is where your database stuff works continue? # whatever here you planned on letting happen. all this is easy and cheap. For your quick demonstration, I suggest setting up asterisk with a dummy interface, downloading the iaxclient and showing that your AGI app would be easy enough to write. You are then only into the project for time, but not any parts. Once you have that down, you would then purchase the parts needed to complete the project from Digium and deploy. If you stick with a T100P interface then you should be able to handle 500 people with 5 minute calls mainly around the business work time and have a small window of safety to not overload the circuits to the point you will have busy signals often. If it is likely you could grow beyond 500 people soon, you may want to buy the T400P card and be able to deploy more digital trunks without taking the system down for more than an asterisk restart. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR only system with scalibility with asterisk???
On Thu, 4 Sep 2003, Paul C wrote: One thing I do find a bit lacking is documentation, but being a developer I know how we all hate writing docs ;-) I've seen the AGI HTML dump, think it makes sense, I guess I just need to do a bit more playing and looking at the examples. use the source, luke, if you know perl, get citats asterisk-perl modules, asterisk.gnuinter.net, they make life really easy digium makes single and quad port E/T1 cards too, very economical, and scale wonderfully well.. Yes, I love the pricing compared to Dialogic cards! When you say they scale well, how many cards are we talking about in a single chassis? well, 2 of the quad-e1 is about as high as i'd go, so 240 tdm channels (and this question has been asked a number of times before , see the varying discourses on the mailing list archives) - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telantek.adsi
On Wed, 2003-09-03 at 22:03, jerk face wrote:
I am working with the telantek.adsi file, and I was
wondering how I would create a softkey for Transfer.
I tried making a key definition and using SENDDTMF
#, but that didn't work. Is there another way I
could do this?
SENDDTMF #
has worked for me once I had enabled the appropriate transfer flag in
the dial statement ('t'). Another approach using flash worked as well:
KEY switch IS Switch OR Switch
FLASH
ENDKEY
Also, does anybody have any ADSI scripts for use with
Asterisk that they would like to share?
I have nothing much, just some trial and error stuff based on the adsi
scripts that come with the adsi source. Mail me if you want to take a
peek.
Something odd I noticed:
SHOWKEYS cwdisable UNLESS nocallwaiting
Does not work within a softkey definition nor do any flag operations. As
I have no access to the adsi specifications I can not tell if this is a
peculiarity of those specs, or a bug in asterisk's implementation of
adsi (adsiprog.c)
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
signature.asc
Description: This is a digitally signed message part
RE: [Asterisk-Users] SIP on TCP
John Todd [EMAIL PROTECTED] writes: I don't know how to automatically determine TCP or UDP for outbound connections without blocking or putting in some really nasty UDP failure detection modes. Maybe this would best be configured to start with just protocol=[tcp,udp] as the only options, but I leave that to the patch writers, as they know far better than me how to make that work. For SIP in general, you look at the transport parameter on the request URI. If it isn't present, you use UDP; otherwise, you use the transport indicated. To do this properly in Asterisk, I would suggest using the same basic mechanism. In other words, to route a call over UDP, you would use: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) Or something like this (syntax needs tweaking): exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED];transport=udp) On the other hand, to route a call over TCP, you would use: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED];transport=tcp) You would want to have the ability to do similar things for sip.conf. For example, you would want the following to be valid: register = [EMAIL PROTECTED];transport=udp/1000 Now, I know that ; is how you start a comment in the Asterisk configuration files, so you'll need to replace it with something else. Comma and | would cause confusion with parameters passed in to Dial()... perhaps ! or #? Finally, you would also want to add a transport parameter to the user sections in sip.conf, like: [2000] type=friend host=dynamic context=international dtmfmode=rfc2833 secret=123456 transport=tcp Unfortunately, I don't have the time to write this patch myself at the moment. I'll note that it's a somewhat nontrivial task, as you have to parse the headers as they arrive to find the Content-Length: or l: header field, find the CR/LF/CR/LF sequence that marks the end of the header, and then count bytes to the end of the message. You can get this right for about 90% of the cases with a quick hack, but getting it right per RFC 3261 is a royal pain. Then there's the whole issue of response routing using the protocol indicated in the topmost Via: header field, and the issue of routing subsequent requests in the same dialog using the Contact: header field... /a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I don't think I understand Call pickup
I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] telantek.adsi
Where is the telantek.adsi file? -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 4:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] telantek.adsi I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF #, but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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] update re. Grandstream + SIP + Echo problems ..
Someone had a patch to retrieve the oldest call from the parking queue... maybe that could help regards Martin On Thu, 4 Sep 2003, WipeOut . wrote: Parking the call is a problem becasue you will not hear the parked call location (because its a blind transfer into the parked call).. The only solution I could think of is to call the person you want to transfer to on the second line, then go back to the first line and blind transfer the call.. (the person you are transfering to will have to hang up after you have spoken to them) What is the process for transfering with the flash button?? I have always used the transfer button and the redial/send button.. no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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 don't think I understand Call pickup
Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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] IVR only system with scalibility with asterisk???
Top posting only: This is great info. A couple of you have already replied with very helpfull and usefull information. Thank you very much!! I am very excited to hear that I can test without purchasing the hardware. I googled and found a IAXClient at http://iaxclient.sourceforge.net/. Is that the program you mean? It looks like * is a very good sofware to pursue and very powerfull (and fairly inexpensive) when hooked up with the Digium cards. I will download and begin trying *. I will likely just place an order for a single analog card just to get the ball rolling very soon. I still would like to hear more about how people are integrating * with external scripts. It was mentioned that the docs may be a little sparse... examples would be GREAT (said in the voice of Tony Tiger). On Thu, Sep 04, 2003 at 01:15:11PM -0500, Steven Critchfield wrote: ...my original post deleted First you need to decide on how many ports you will need, how important ease of scalability is. For the number of ports, you need to decide how much tolerance you have for the people remotely to deal with a busy signal. So far you mentioned 45 people making 3-4 calls a day over a ~8 hour day. The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and everyone waited till it became available. My guess is you don't want people on redial that often and waiting for the port to come open. Next, you move on to what is the acceptable idle amount of service available. If you scaled up to say 5 lines, and the call length is short, then you will have your service mostly idle, but it can handle peak times better. I'll let you continue this line of questioning internally. Next to decide on hardware, if you think you may need more than 10 lines, you need to move to digital trunks. You can start with a T100P and a channel bank until your costs justify switching over to a T1. The benefit is already having the hardware in hand and used to it while on spending a little more short term to get the FXO channel bank that you will either sell off later, or convert to FXS for internal extensions if you want to switch services. If you already have a PBX in house and can drop a T1 interface to you asterisk box, that is good too. As for your application. You mention looking into perl modules, so I assume you have some perl familiarity. From AGI you can script up any database access and prompting you so wish to undertake. Essentially it will come out to be something like. stream file(prompt) while (not enough digits) wait for digits collect dialed digits validate(digits) # in this sub is where your database stuff works continue? # whatever here you planned on letting happen. all this is easy and cheap. For your quick demonstration, I suggest setting up asterisk with a dummy interface, downloading the iaxclient and showing that your AGI app would be easy enough to write. You are then only into the project for time, but not any parts. Once you have that down, you would then purchase the parts needed to complete the project from Digium and deploy. If you stick with a T100P interface then you should be able to handle 500 people with 5 minute calls mainly around the business work time and have a small window of safety to not overload the circuits to the point you will have busy signals often. If it is likely you could grow beyond 500 people soon, you may want to buy the T400P card and be able to deploy more digital trunks without taking the system down for more than an asterisk restart. -- 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 don't think I understand Call pickup
What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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] Re: Asterisk Jitters
Message: 3
Subject: Re: [Asterisk-Users] Re: Asterisk Jitters
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 03 Sep 2003 23:13:58 -0500
Reply-To: [EMAIL PROTECTED]
On Thu, 2003-09-04 at 01:43, Zak wrote:
I have three fxos from Digium installed in the box.
The Box got Pentium 4 2.4 Ghz and 512 RAM.
I had the box working fine once but it stopped working (jitters) after
a reboot.
Did you make sure the zap card drivers are loaded? Have you checked your
IRQs to make sure they didn't wander and start causing problems?
Steven,
The zap card drivers are loaded correctly and there don't seem to be
any IRQ problem.
Here is what I get from /proc/interrupts.
I'm not sure the sound card and eth0 using the same IRQ could be
causing the problem?
CPU0
0: 17728993 XT-PIC timer
1: 44037 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 9776462 XT-PIC eth0, Intel ICH2
8: 1XT-PIC rtc
9: 103179821 XT-PIC wcfxo
10: 217202079 XT-PIC nvidia, wcfxo, wcfxo
12: 524504 XT-PIC PS/2 Mouse
14: 165140 XT-PIC ide0
15: 281208 XT-PIC ide1
NMI: 0
ERR: 0
Zak
If you have one, and the card is up and running, then it would be used
for timing. Basically it is just needed in this case to make sure
asterisk keeps chugging along at a known speed.
What speed hardware are you using?
Date: Wed, 03 Sep 2003 21:05:04 -0700
From: Zak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk Jitters
Reply-To: [EMAIL PROTECTED]
Hi Steven,
I have a zap device installed in the box but I'm not sure if that's the one used for timing.
thanks.
Zak
Subject: Re: [Asterisk-Users] Asterisk Jitters
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 03 Sep 2003 11:17:15 -0500
Reply-To: [EMAIL PROTECTED]
Do you have a zap device for timing?
On Wed, 2003-09-03 at 17:48, Zak wrote:
Hi,
Every time I dial into my asterisk box i hear nothing but asterisk
jittering.
The following is an example of what I get on the asterisk CLI
Thanks
*CLI DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user
'xirak' is 1
out of 0
DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route:
Contact hop
: sip:192.168.7.3
-- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack
DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format
changed from U
NKN to ULAW
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 16
0 sample intervals
-- Playing 'vm-login'
DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping
retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 0
sample intervals
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 0
sample intervals
WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain):
Couldn't read u
sername
== Spawn extension (extensions, 1001, 1) exited non-zero on
'SIP/xirak-259d'
DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak)
RE: [Asterisk-Users] I don't think I understand Call pickup
Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Jitters
On Thu, 4 Sep 2003, Zak wrote: The zap card drivers are loaded correctly and there don't seem to be any IRQ problem. Here is what I get from/proc/interrupts. I'm not sure the sound card and eth0 using the same IRQ could be causing the problem? 9: 103179821XT-PIC wcfxo 10: 217202079 XT-PIC nvidia, wcfxo, wcfxo don't know if its directly related, but don't share IRQ on the two wcfxo and the nvidia, AND don't use XT-PIC, APIC is much, much better 16: 28988582 IO-APIC-level wcfxo 17: 28988561 IO-APIC-level wcfxo -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
WipeOut . wrote: now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. Mmmmh, Does not ADSL stands for ASIMETRIC Digital Subscriber Line ? So if its a 128 Kbps ADSL... what is your Asimetry relation ? You might be having 128 Kbps in one way but 64 Kbps ( Or less ) on the other... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR only system with scalibility with asterisk???
On Thu, 2003-09-04 at 14:10, PJ Welsh wrote: Top posting only: This is great info. A couple of you have already replied with very helpfull and usefull information. Thank you very much!! I am very excited to hear that I can test without purchasing the hardware. I googled and found a IAXClient at http://iaxclient.sourceforge.net/. Is that the program you mean? yes. While it isn't ready for prime time use, it is a easy way to start it up from a windows or OSX system. It works much easier than a SIP or H323 soft phone. It looks like * is a very good sofware to pursue and very powerfull (and fairly inexpensive) when hooked up with the Digium cards. I will download and begin trying *. I will likely just place an order for a single analog card just to get the ball rolling very soon. If you read through the archive you will find that an analog card is prone to signaling problems. Basically it is not something you would want to depend upon for a business. Not to mention that is is 1/5th of the cost of a T100P card that you will probably need to move up to in the short term to handle the load you will generate during parts of your day. I still would like to hear more about how people are integrating * with external scripts. It was mentioned that the docs may be a little sparse... examples would be GREAT (said in the voice of Tony Tiger). There is quite a bit available via the archive. Not to mention there are a few people on this list that are selling services based on asterisk and may for a small fee give you a great jump start on hardware and/or software setup/creation. If you ask directly for this help, I'm sure someone will contact you shortly to offer assistance. On Thu, Sep 04, 2003 at 01:15:11PM -0500, Steven Critchfield wrote: ...my original post deleted First you need to decide on how many ports you will need, how important ease of scalability is. For the number of ports, you need to decide how much tolerance you have for the people remotely to deal with a busy signal. So far you mentioned 45 people making 3-4 calls a day over a ~8 hour day. The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and everyone waited till it became available. My guess is you don't want people on redial that often and waiting for the port to come open. Next, you move on to what is the acceptable idle amount of service available. If you scaled up to say 5 lines, and the call length is short, then you will have your service mostly idle, but it can handle peak times better. I'll let you continue this line of questioning internally. Next to decide on hardware, if you think you may need more than 10 lines, you need to move to digital trunks. You can start with a T100P and a channel bank until your costs justify switching over to a T1. The benefit is already having the hardware in hand and used to it while on spending a little more short term to get the FXO channel bank that you will either sell off later, or convert to FXS for internal extensions if you want to switch services. If you already have a PBX in house and can drop a T1 interface to you asterisk box, that is good too. As for your application. You mention looking into perl modules, so I assume you have some perl familiarity. From AGI you can script up any database access and prompting you so wish to undertake. Essentially it will come out to be something like. stream file(prompt) while (not enough digits) wait for digits collect dialed digits validate(digits) # in this sub is where your database stuff works continue? # whatever here you planned on letting happen. all this is easy and cheap. For your quick demonstration, I suggest setting up asterisk with a dummy interface, downloading the iaxclient and showing that your AGI app would be easy enough to write. You are then only into the project for time, but not any parts. Once you have that down, you would then purchase the parts needed to complete the project from Digium and deploy. If you stick with a T100P interface then you should be able to handle 500 people with 5 minute calls mainly around the business work time and have a small window of safety to not overload the circuits to the point you will have busy signals often. If it is likely you could grow beyond 500 people soon, you may want to buy the T400P card and be able to deploy more digital trunks without taking the system down for more than an asterisk restart. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] telantek.adsi
It's my asterisk.adsi file that I changed to suit my needs. I was just looking at the file name and not thinking while I was typing the email. --- Wade J. Weppler [EMAIL PROTECTED] wrote: Where is the telantek.adsi file? -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 4:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] telantek.adsi I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF #, but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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 __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Jitters
On Thu, 2003-09-04 at 22:13, Zak wrote: Message: 3 Subject: Re: [Asterisk-Users] Re: Asterisk Jitters From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 03 Sep 2003 23:13:58 -0500 Reply-To: [EMAIL PROTECTED] On Thu, 2003-09-04 at 01:43, Zak wrote: I have three fxos from Digium installed in the box. The Box got Pentium 4 2.4 Ghz and 512 RAM. I had the box working fine once but it stopped working (jitters) after a reboot. Did you make sure the zap card drivers are loaded? Have you checked your IRQs to make sure they didn't wander and start causing problems? Steven, The zap card drivers are loaded correctly and there don't seem to be any IRQ problem. Here is what I get from/proc/interrupts. I'm not sure the sound card and eth0 using the same IRQ could be causing the problem? CPU0 0: 17728993 XT-PIC timer 1: 44037XT-PIC keyboard 2: 0XT-PIC cascade 5:9776462 XT-PIC eth0, Intel ICH2 8: 1XT-PIC rtc 9: 103179821XT-PIC wcfxo 10: 217202079 XT-PIC nvidia, wcfxo, wcfxo Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same interupt. I'd say try removing a 2 WCFXO cards from the system and see if the interupts free up, and your jitter stops. 12: 524504 XT-PIC PS/2 Mouse 14: 165140 XT-PIC ide0 15: 281208 XT-PIC ide1 NMI: 0 ERR: 0 Zak If you have one, and the card is up and running, then it would be used for timing. Basically it is just needed in this case to make sure asterisk keeps chugging along at a known speed. What speed hardware are you using? Date: Wed, 03 Sep 2003 21:05:04 -0700 From: Zak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk Jitters Reply-To: [EMAIL PROTECTED] Hi Steven, I have a zap device installed in the box but I'm not sure if that's the one used for timing. thanks. Zak Subject: Re: [Asterisk-Users] Asterisk Jitters From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 03 Sep 2003 11:17:15 -0500 Reply-To: [EMAIL PROTECTED] Do you have a zap device for timing? On Wed, 2003-09-03 at 17:48, Zak wrote: Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user 'xirak' is 1 out of 0 DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop : sip:192.168.7.3 -- Executing VoiceMailMain2(SIP/xirak-259d, ) in new stack DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from U NKN to ULAW DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 16 0 sample intervals -- Playing 'vm-login' DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 0 sample intervals DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 0 sample intervals WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): Couldn't read u sername == Spawn extension (extensions, 1001, 1) exited non-zero on 'SIP/xirak-259d' DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak) -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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] Traffic Modelling (was IVR only system with scalibility...)
The question was posed: incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours The comment was made (taken out of context): The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and everyone waited till it became available. Unfortunately as we all know, asking callers to guess when the line is free and equally spacing their calls is not terribly realistic (as the author of the comment above goes on to imply). So how does one analyse such a situation? Using statistical traffic modelling! For more information, see http://www.erlang.com/calculator/erlb/ Plug in: Busy Hour Traffic: 0.937 Erlangs (based on 45 * 4 * 2.5 / 480) Acceptable Blocking Factor: 1% (we will accept 1 in 100 calls receiving a busy signal) Result: you will need 5 incoming lines. If you are willing to tolerate (say) 3% of calls receiving a busy signal, you can get by with 4 lines etc. and etc. Hope you find the above useful in planning your Asterisk installation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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] update re. Grandstream + SIP + Echo problems ..
it's 512k/128k actually ... :) Dave - Original Message - From: Ing. Angel Gomez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 9:23 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. WipeOut . wrote: now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. Mmmmh, Does not ADSL stands for ASIMETRIC Digital Subscriber Line ? So if its a 128 Kbps ADSL... what is your Asimetry relation ? You might be having 128 Kbps in one way but 64 Kbps ( Or less ) on the other... ___ 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 don't think I understand Call pickup
Oh and with the recent CVS code call pickup is broken for sip phones ... I just got that from bugtracker Martin On Thu, 4 Sep 2003, Martin Pycko wrote: You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
When you modify the Zap config you have to stop/start asterisk or do a restart (not reload) On Thu, 2003-09-04 at 14:25, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Ahh... Now it is working, but the phone which is ringing keeps on ringing after the pickup (and I have a connection between the zap and sip channel). -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:59 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I don't think I understand Call pickup
Ok, explains why the phone keeps ringing then -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 22:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Oh and with the recent CVS code call pickup is broken for sip phones ... I just got that from bugtracker Martin On Thu, 4 Sep 2003, Martin Pycko wrote: You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup = callgroup of the sip phones Hmm...I must be stupid ;O), can't get it to work. In zapata.conf I give the parameter pickupgroup=1 (which covers my 15 zap channels) and in sip.conf I give the parameter callgroup=1 on the phone I want to be able to pick up. Is this right, or have I misunderstood it completely? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] I don't think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything in sip.conf or is it enough to specify it in zapata.conf? -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: 4. september 2003 21:08 To: Asterisk maillist (E-mail) Subject: Re: [Asterisk-Users] I don't think I understand Call pickup Lets say that you have two phones: Zap/1 and Zap/2 and there comes a call over IAX to Zap/1 since channel 1 is in the callgroup 1 and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a Nothing to pick up answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ 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 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 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] SIP on TCP
At 1:28 PM -0500 9/4/03, Adam Roach wrote: John Todd [EMAIL PROTECTED] writes: I don't know how to automatically determine TCP or UDP for outbound connections without blocking or putting in some really nasty UDP failure detection modes. Maybe this would best be configured to start with just protocol=[tcp,udp] as the only options, but I leave that to the patch writers, as they know far better than me how to make that work. For SIP in general, you look at the transport parameter on the request URI. If it isn't present, you use UDP; otherwise, you use the transport indicated. To do this properly in Asterisk, I would suggest using the same basic mechanism. In other words, to route a call over UDP, you would use: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) Or something like this (syntax needs tweaking): exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED];transport=udp) On the other hand, to route a call over TCP, you would use: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED];transport=tcp) You would want to have the ability to do similar things for sip.conf. For example, you would want the following to be valid: register = [EMAIL PROTECTED];transport=udp/1000 Now, I know that ; is how you start a comment in the Asterisk configuration files, so you'll need to replace it with something else. Comma and | would cause confusion with parameters passed in to Dial()... perhaps ! or #? Finally, you would also want to add a transport parameter to the user sections in sip.conf, like: [2000] type=friend host=dynamic context=international dtmfmode=rfc2833 secret=123456 transport=tcp Unfortunately, I don't have the time to write this patch myself at the moment. I'll note that it's a somewhat nontrivial task, as you have to parse the headers as they arrive to find the Content-Length: or l: header field, find the CR/LF/CR/LF sequence that marks the end of the header, and then count bytes to the end of the message. You can get this right for about 90% of the cases with a quick hack, but getting it right per RFC 3261 is a royal pain. Then there's the whole issue of response routing using the protocol indicated in the topmost Via: header field, and the issue of routing subsequent requests in the same dialog using the Contact: header field... /a I should get some more sleep and remember my RFC's a bit more. Of course, as you say, there is a protocol specified in the request which can be used to determine the correct response. (What happens when the protocol in the request doesn't match the one actually used to deliver the request? woo hoo!) Could the patch simply look at the Via: header and grab the the last protocol? Probably not. Pesky RFC compliance... Out of curiosity, why would you need to parse through the whole header to find that data? Looks like it's right in the Via: header. I am uncertain if the transport protocol should be able to be specified in the register or Dial portions of sip.conf and extensions.conf, respectively. I am happier leaving that determined in the definition of the SIP peer, and not flexible on a per-call basis, just like most of the other features that are defined in sip.conf. If you need per-call flexibility, use a different SIP peer that gets built with includes, if they are added to sip.conf (see my long comments from earlier today on revamping sip.conf) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] telantek.adsi
I figured. Any new ADSI scripts always peak my interest. :) Telantek? Are you from Telantek in Toronto? -wade -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] Sent: Thursday, September 04, 2003 3:29 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] telantek.adsi It's my asterisk.adsi file that I changed to suit my needs. I was just looking at the file name and not thinking while I was typing the email. --- Wade J. Weppler [EMAIL PROTECTED] wrote: Where is the telantek.adsi file? -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 4:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] telantek.adsi I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF #, but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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 __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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] SIP on TCP
John Todd [mailto:[EMAIL PROTECTED] writes: Out of curiosity, why would you need to parse through the whole header to find that data? Looks like it's right in the Via: header. I was talking about message framing. I'll rephrase in the form of a FAQ. Q: How do you know where the SDP (or any other body) ends? A: You have to find and parse the Content-Length header field, find the end of the headers, and count bytes. /a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Packet8 DTA310
Asterisk recognizes and interprets the XML correctly? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Tuesday, September 02, 2003 12:26 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Packet8 DTA310 Well this debug desn't show the bad call setup. And furthermore all commands are accepted by the asterisk/UA. Martin On Mon, 1 Sep 2003, Andrew Joakimsen wrote: There might be some other stuff mixed in there as well, 64.36.104.205 is asterisk and 64.36.104.206 is the DTA 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 From: asterisk sip:[EMAIL PROTECTED];tag=as17328ab1 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/1 (no NAT) to 64.36.104.203:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 From: asterisk sip:[EMAIL PROTECTED];tag=as17328ab1 To: sip:[EMAIL PROTECTED];tag=6b0caf74-4936-bceb-18dc-3ee5469aa3f6 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Grandstream SIP UA 1.0.3.81 Contact: sip:[EMAIL PROTECTED];user=phone Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 10 headers, 0 lines Sip read: SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport From: sip:[EMAIL PROTECTED];tag=t2d9e0a11a85c88g To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 100 SUBSCRIBE Contact: sip:[EMAIL PROTECTED] Expires: 3600 Max-Forwards: 70 Event: traverse User-Agent: DTA SIP/0.11.8 NNOS/VR30 Content-Type: application/sdp Content-Length: 156 v=0 o=0403532579 0 0 IN IP4 64.36.104.206 =-m3*CLI c=IN IP4 64.36.104.206 t=0 0 m=audio 8002 RTP/AVP 18 101 a=ptime:10 a=rtpmap:101 telephone-event/8000 13 headers, 8 lines Using latest SUBSCRIBE request as basis request Sending to 64.36.104.206 : 5060 (non-NAT) Looking for in international Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport From: sip:[EMAIL PROTECTED];tag=t2d9e0a11a85c88g To: sip:[EMAIL PROTECTED];tag=as57545bcd Call-ID: [EMAIL PROTECTED] CSeq: 100 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Content-Length: 0 to 64.36.104.206:5060 Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 From: sip:[EMAIL PROTECTED];tag=as57545bcd To: sip:[EMAIL PROTECTED];tag=t2d9e0a11a85c88g Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Content-Type: application/xpidf+xml Content-Length: 352 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id= address uri=sip:[EMAIL PROTECTED];user=ip priority=0,80 status status=open / msnsubstatus substatus=online / /address /atom /presence (no NAT) to 64.36.104.206:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 From: sip:[EMAIL PROTECTED];tag=as57545bcd To: sip:[EMAIL PROTECTED];tag=t2d9e0a11a85c88g Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Server: DTA SIP/0.11.8 NNOS/VR30 Content-Length: 0 8 headers, 0 lines Message is NOTIFY hm3*CLI -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Saturday, August 30, 2003 12:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Packet8 DTA310 Post the sip debug .. maybe someone will help you. Martin On Sat, 30 Aug 2003, Andrew Joakimsen wrote: Has anyone been successful in using the DTA310 as provided by Packet8 to work with asterisk? I have gotten it to register with Asterisk but whenever I try to dial a call all I get is silence, when I dial an invalid extension I get a fast busy signal. When looking at the SIP debug it seems that it is transmitting XML. ___ 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]
Re: [Asterisk-Users] Traffic Modelling (was IVR only system with scalibility...)
Nice goin'!! I will use this for a reference point to establish baseline numbers of phone lines. The good news is that the equation is not linear (eg 45 people need 5 lines, 100 need 10 lines). So I can double my potential users and ONLY need 2 more lines (qty 7). The bad news is that I don't 100% think I will have purely random connections. On Thu, Sep 04, 2003 at 12:48:58PM -0700, George Pajari wrote: The question was posed: incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours The comment was made (taken out of context): The quick math says that 45 people with 4 calls is 180 calls a day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could handle the load if the call was under 2.5 minutes long and everyone waited till it became available. Unfortunately as we all know, asking callers to guess when the line is free and equally spacing their calls is not terribly realistic (as the author of the comment above goes on to imply). So how does one analyse such a situation? Using statistical traffic modelling! For more information, see http://www.erlang.com/calculator/erlb/ Plug in: Busy Hour Traffic: 0.937 Erlangs (based on 45 * 4 * 2.5 / 480) Acceptable Blocking Factor: 1% (we will accept 1 in 100 calls receiving a busy signal) Result: you will need 5 incoming lines. If you are willing to tolerate (say) 3% of calls receiving a busy signal, you can get by with 4 lines etc. and etc. Hope you find the above useful in planning your Asterisk installation. ___ 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] frames/packet
Sure you get nearly your line rate on the line from your house to your DSLAM. Once you get to the DSLAM, you are mixing with all the others users traffic. Some DSL providers cap your guaranteed bandwidth at about 10kbps. Lee Goodman - Original Message - From: Timothy Soos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 5:15 PM Subject: Re: [Asterisk-Users] frames/packet -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Don't count on being able to push the full 32 simultaneous calls. Cable modems are notorious for failing to deliver the full bandwidth the subscriber is paying for. In many places customers end up getting 33Kbbps or less on a regular basis. The available bandwidth tends to fluctuate frequently. In contrast, many (if not most) DSL systems will provide a constant bandwidth close to the maximum rate the customer is paying for. Most DSL providers will cap the available CONTINUOUS rate to around 540Kbps for a 650Kbps connection (as an example). All DSL providers will allow the rate to momentarily spike to the full speed the customer is paying for. A small number of DSL providers allow the full bandwidth to be used all the time. Tim On Wednesday 03 September 2003 10:15 am, Paul Lambert wrote: Not yet. implies that it is coming. I know it would help on Internet connections such as fixed wireless and cable modem where packet rate is an issue. 20ms translates to 50 packets/sec. I believe cable modem upstream packet rates cap at 150-200 packets/sec. G729 gets the bit rate down to 8kbits. So based on a bit rate of 256K the theory is that the link could handle 32 calls. But, that would produce packets coming out at a rate of 1600 packets/sec beyond the limitation of most Internet connections including a T1. Martin Pycko wrote: Not yet. Asterisk always sends 20 ms of voice data per packet. regards Martin On Wed, 3 Sep 2003, Paul Lambert wrote: Noticed that I can adjust the number if frames/packet on the GrandStream phone. Can * do the same? ___ 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 - -- Thanks, Timothy Soos XQL, LLC 303-480-8228 720-979-3128 (Direct) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/VloOnG4mv6BHya0RAskBAJ9va4sGW2ExToy+N/8X8Q4miVE8KwCfUkv2 i2kbOgRzddeWpVmZ00/+2pw= =ErZZ -END PGP SIGNATURE- ___ 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] Pointer to upgrade 7960sip beyond v3.2.0?
Joe, Finally figured some of the 7960 stuff out. The **# only works with early code releases. Somewhere around v3-to-v4 that was discontinued, and replaced with that password option to unlock the front panel instead. I've can now get to the net sip configuration panel, and I've got SIPDefault.cnt file that gets loaded at boot time. One of two 7960's is upgraded to v4.4, however the second one constantly reboots (only with v4.4) and never gets to a usable status. Using a sniffer trace to observe, the reboot happens after dhcp is proper and SIPdefault is read by the phone (via tftp). Within milliseconds of reading that file, the phone reboots, then repeats the same process. The first phone boots v4.4 correctly every time. Late yesterday, I removed the statements in SIPdefault after comments relative to ...added in Release 3.1 and the unstable phone will bootup and stay working on v4.4. If I add those config statements, the constant reboot happens again. So, it almost looks like there might be a memory issue (as in total mem comsumption can't exceed some value, or it reboots), but that is a total guess. Is there an administrators or diagnostic manual fot the 7960sip, or has cisco left it up to all of us to figure out how to diagnose strange problems like this? I can't seem to find much of anything at the cisco site, nor through google. I'm somewhat reluctant to try v5.4 code given the warning that one can't go back. Any help or suggestions would be greatly appreciated. Rich Rich, you can do **# and go into the Network config and hit **# again, you should notice the LockPad come unlocked and then you can make changes. If you upgraded, the default password is cisco Joe -Original Message- Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't unlock it to do any configs. Any thoughts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 backup proxy registration
Hi, I'm sorry to ask this question, but I thought I'd rather ask it here before messing up with cisco. Is anybody running cisco 7960 in redundant configuration? I mean I want the phone to be registered with both primary and backup proxy (asterisks) so that service continues to work in case of primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10 proxy1_port: 5060 proxy_backup: 192.168.1.12 proxy_backup_port: 5060 The problem is that 7960 registers all the configured lines with primary proxy, but the line 1 only with backup proxy. It's not about registration failure. The phone doesn't even try to register other 5 lines. As a result if the primary proxy fails incoming calls work for the line 1 only. Has anybody managed to register all the lines with backup proxy? I'm running software 4.4, the last version before digital signature was introduced. Should I upgrade? Or may be I'm missing something in configuration? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modems and Tivos? Oh my!
Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Also, on a somewhat unrelated question: How does the FXS card generate ringing voltages if the PC only supplies 12 volts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modems and Tivos? Oh my!
On Thu, 2003-09-04 at 17:22, Peter Pauly wrote: Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Why would one use dialup for a TiVo. My TiVo has never touched a telephone line ever. When I bought it I hacked it to work over my cable modem link using PPP to my workstation and have never risked loosing the internal hardware. Also, on a somewhat unrelated question: How does the FXS card generate ringing voltages if the PC only supplies 12 volts? Probably with similar laws of electricity that allows car spark plugs to operate in excess of 50K volts when the battery only supplies 12volts, or that allow flourescent lights to run at a few k volts while the wall puts out 110 or 220. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk stops responding
It happened once again here. This time I called an IVR (SIP to SIP) and upon sending the 1st DTMF tone, * bombed out. The console got filled with these messages (and they wouldn't stop): DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again.. * stopped responding and I had to kill the process manually. *CLI show version Asterisk CVS-08/22/03-22:24:05 built by [EMAIL PROTECTED] on a i686 running Linux Has anybody else seen this message? Regards, Andres On Thursday 28 August 2003 13:37, Andres wrote: We run Iptel's SER as our SIP Server. All subs register with our SIP Server, but if anyone needs to call the PSTN then the call gets forwared to *. The Request to schedule in the past messages have to do with MOH and I was told it was due to a slow PC. I don't think it is related with Asterisk hanging up. Regards, Andres On Thursday 28 August 2003 13:27, David Harris wrote: Gazing at the console I was able to determine the exact time Asterisk froze. Even with DEBGUG on it did not show anything important. The moment it freezes is when a call from Phone1 tries to connect to a SIP Provider like Iconnect: I have not been able to pin point exactly what event causes the freeze-up but I have been on the console when it has happened. It didn't print out anything interesting. The call I was on cut off. Phone1Our SIP Server---Our AsteriskSIP Provider It was by no means 100% reproducible. Maybe 1 out of 10 calls caused the trouble. Same here except I would say more like 1 out of 100 calls. A bad symptom would be that the command show sip channels would show several calls, even though they had hungup a long time ago. I definitely have this problem. Troubleshooting revealed that the BYE message was not being sent by our SIP Server to the Asterisk server upon hangup. We rectified this and we no longer see those phantom SIP Channels and Aterisk has not froze for about a week. What is your SIP Server what does it do? Maybe I have the same issue with my Cisco Voice Gateway not sending the BYE message sometimes. But would this cause asterisk to freeze? Other symptoms I have are these errors in the asterisk messages log file Aug 27 09:21:00 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:24 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:29 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:35 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:22:05 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Thanks, David Harris ___ 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] Modems and Tivos? Oh my!
On Thu, 2003-09-04 at 17:22, Peter Pauly wrote: Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Assuming that you can do native zaptel bridging (Going from an FXS port to an FXO port in the same machine), you should be able to get up to 33.6. No 56k, unfortunately, because of the multiple D/A A/D conversions. If you're codecing the audio and passing it over IP, you should be able to get 33.6 if you use ulaw (non-compressed) encoding. Any of the compression based codecs will most likely make your modem link up at 9600 or a flaky 14.4. Why would one use dialup for a TiVo. My TiVo has never touched a telephone line ever. When I bought it I hacked it to work over my cable modem link using PPP to my workstation and have never risked loosing the internal hardware. Or spend a few bucks and get yourself one of the ethernet card kits. Then you don't have to worry about ppp connections and you can drag the video off the unit as well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
