RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guenther Boelter I have 3 Grandstream Budge-Tone 100 with Firmware 1.0.7.11beta, and they are working very well since more then 4 month now. I'm using two Grandstream Budgetone 101 without problems. Only problem is the lackness of alphanumeric display and distinctive ring. Where can I find this updated Budgetone firmware? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sharing a dialplan
Hi, I need to connect two sites and two Asterisk servers sharing their dialplan. In fact users usually can be moved at different offices and carry their phone number. What's the best way to do this? - switch statement - DUNDI ? Thanks for any help Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID over analog?
Hello, I agree with Damon's comments below. Just for information. Eicon do have the Diva Server Analog range of cards that will work with asterisk. You can plug these into Analog lines and then use them with Asterisk via the CAPI interface of the Diva Server driver. If you have CLIP (The Caller ID service) on the analog line you can get the calling party number indicated to asterisk. It is also possible to assign a number to each line in the driver, that will be shown as the Called Party Number in Asterisk. However, it would be better to get your T1 working. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Damon Estep Sent: 30 January 2006 18:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DID over analog? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Monday, January 30, 2006 9:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DID over analog? I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken One would have to think that fixing the T1 issue is a far better solution, have you tried asking the questions related to the T1 fax problems? Analog DID trunks are problematic at best, and not supported as far as I have seen in asterisk. Most reliable DID trunks are 4 wire, not 2 wire. They require a special DID trunk interface and I have not seen one for asterisk. While there are 2 wire DID trunks form some telcos, they are a joke. ISDN BRI (2B+D) is also a viable solution as multiple numbers can be routed to an ISDN BRI line - call your telco and see if they will do multiple numbers on ISDN and then look at the capi cards for use with asterisk like the DIVA series from EICON. I have no personal experience with them but many others do. Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Kirk IP600
I'm going to try, Thanks very much Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Remco Barende Inviato: lunedì 30 gennaio 2006 20.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Kirk IP600 Hi! Yes, it works (sort of) but I still have some issues. When using more than 2 handsets some of them do not always ring on an incoming call. This might be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's the driver I created a howto for it, you can find it here: http://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt Let me know if you find any errors / omissions, or the solution to the ringing problem :) On Mon, 30 Jan 2006, Giordano Grandis wrote: Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forward a call from AGI/PHP script
Any suggestions on how to go about this? so person calls, recording: "press2 to call cell phone", user presses 2, call forwards to my cell phone. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Try setting the Callerpresentation to something else: http://www.voip-info.org/wiki/page_history.php?page_id=1682preview=2 SetCallerPres(prohib) actually worked! Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interface card for Euro-ISDN (BRI)
Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdrtool
Hello, Call sip:[EMAIL PROTECTED] Regards harry --- Jimmy Smith [EMAIL PROTECTED] a écrit : anyone having weird problems on latest cdrtool? #!/usr/bin/php4 *Fatal error*: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in */var/www/CDRTool/SOAP/client_lib.php*on line *2 always get weird error like that * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward a call from AGI/PHP script
execute the dial command from AGI. e.g. exec(dial(SIP/provider/2394892348)) you may want to reset or fork the cdr so you can have the record for the IVR interaction and a different record for the call you are connecting. See ForkCDR and ResetCDR hope this helps, Cristi On 1/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Any suggestions on how to go about this? so person calls, recording: press 2 to call cell phone, user presses 2, call forwards to my cell phone. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
Look here for the updated firmware: http://www.grandstream.com/BETATEST/ Don't ask me why, but you really have to use capital-letters for the word BETATEST!! If you are interested in 1.0.7.11beta, i can gsend you a copy via email because it's not on the server anymore. Guenther Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guenther Boelter I have 3 Grandstream Budge-Tone 100 with Firmware 1.0.7.11beta, and they are working very well since more then 4 month now. I'm using two Grandstream Budgetone 101 without problems. Only problem is the lackness of alphanumeric display and distinctive ring. Where can I find this updated Budgetone firmware? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On 31 Jan 2006, at 09:12, John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. I'm very happy with an Eicon Diva Server V-BRI that I bought a couple months ago. The only drawback is that it doesn't do any fax traffic apparently. It works with chan_capi-cm from Sourceforge. http://www.eicon.com/worldwide/products/MediaGateways/diva-server- vbri.htm jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_snmp
Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] file.c:509 ast_openstream_full: File 100 does not exist in any format
Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards Harry Jan 29 14:34:43 WARNING[2568]: pbx.c:2403 __ast_pbx_run: Timeout, but no rule 't' in context 'info' -- Executing Answer(SIP/86-a9b4, ) in new stack -- Executing Queue(SIP/86-a9b4, info|tn||100) in new stack -- Started music on hold, class 'default', on channel 'SIP/86-a9b4' -- outgoing agentcall, to agent '101', on 'Local/[EMAIL PROTECTED],1' -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack -- Called Agent/101 -- Agent/101 answered SIP/86-a9b4 Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory -- Stopped music on hold on SIP/86-a9b4 -- Executing Dial(Local/[EMAIL PROTECTED],2, Sip/85|30|t) in new stack -- Called 85 -- SIP/85-7874 is ringing == Spawn extension (support, info, 2) exited non-zero on 'SIP/86-a9b4' == Spawn extension (info, 85, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Kirk IP600
I installed the chan_sccp and configured the sccp.conf, but when try to start asterisk I get this error [chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module chan_sccp.so failed! Thanks for all Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Remco Barende Inviato: lunedì 30 gennaio 2006 20.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Kirk IP600 Hi! Yes, it works (sort of) but I still have some issues. When using more than 2 handsets some of them do not always ring on an incoming call. This might be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's the driver I created a howto for it, you can find it here: http://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt Let me know if you find any errors / omissions, or the solution to the ringing problem :) On Mon, 30 Jan 2006, Giordano Grandis wrote: Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
HI, all newer HFC-S cards will do. Depending on your application and system, you could easily ebaying an used Fritz!Card PCI or some active AVM B1 controller. Depending on the card you want to use you must se ZAPHFC or mIISDN/chan_isdn or chan_capi or mixtures with 2 different cards ... good luck, but there are enough HowTos available ... --Juergen Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Web interface
+++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. -- regards Vikram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Budgetone mass deployment?
PB == Phil Blundell [EMAIL PROTECTED] writes: PB Right now I'm still using their Java thing, but it's slow enough PB that one of these days I guess I'll crack and reimplement that PB stuff directly in python. I think the algorithm is described on PB the voip-info.org wiki someplace. A trick about the java thing: It actually runs in gij. I modified the invocation script to go like this -- 8 -- #!/bin/bash GAPSLITE_HOME=/usr/local/lib/grandstream-encode # Do NOT modify below this line LD_LIBRARY_PATH=$LD_LIBRARY_PATH:/usr/local/lib:$GAPSLITE_HOME/lib/`uname -m` export LD_LIBRARY_PATH gij -classpath $GAPSLITE_HOME/gapslite.jar:$GAPSLITE_HOME/bcprov-jdk14-124.jar:$GAPSLITE_HOME com.grandstream.cmd.TextEncoder $* -- 8 -- /usr/local/lib/grandstream-encode is a directory containing the various jar files that Grandstream provides. I have found that gij improves the speed enough to make it viable for us. If you still find it too slow, the jar files can most likely be compiled with gcj, thereby avoiding the slow startup. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Default value for ASTERISK_VERSION_NUM
I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default value is 00. I thought the value should be 010200. I know many people have problems compiling chan_bluetooth because of this inconsistency. Anyone has the last word on this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On Tue, 31 Jan 2006, Jens Vagelpohl wrote: On 31 Jan 2006, at 09:12, John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. I'm very happy with an Eicon Diva Server V-BRI that I bought a couple months ago. The only drawback is that it doesn't do any fax traffic apparently. It works with chan_capi-cm from Sourceforge. The 'V' version of that card is for (V)oice. The standard BRI do support Fax/analog Modem and even RTP with codecs and anti-jitter (echo-cancel too). I'm currently working on support for this CAPI-RTP with chan_capi-cm. Armin http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vbri.htm jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] missing pre pattern matching feature
Hi,
is there a way to executing commands in the dialplan regardless which number is
dialed before
the pattern matching starts ?
when a call enters the first context it would be nice if i can set some
variable or manipulate
a callerid, or what ever before the patternmatching starts.
a solution like this:
[firstcontext]
exten = _.,1,Set(usergroup=1)
exten = _.,2,Goto(firstcontext-post,${EXTEN},1)
[firstcontext-post]
exten = _1.,1,NoOp()
exten = _2.,1,NoOp()
exten = _3.,1,NoOp()
The pattern ._ brings up a warning not to use it.
Is there a save way to workaround this problem without adding macros before
each pattern ?
a predefined channel variable which preserves the starting context would also
be nice.
Harald Holzer
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Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On 31 Jan 2006, at 10:06, Armin Schindler wrote: I'm very happy with an Eicon Diva Server V-BRI that I bought a couple months ago. The only drawback is that it doesn't do any fax traffic apparently. It works with chan_capi-cm from Sourceforge. The 'V' version of that card is for (V)oice. The standard BRI do support Fax/analog Modem and even RTP with codecs and anti-jitter (echo- cancel too). I'm currently working on support for this CAPI-RTP with chan_capi-cm. Yes, I bought it specifically for the Voice optimizations - but my impression was that this was an optimization that would retain other, more basic functions like handling Fax ;) jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Monday 30 January 2006 21:48, [EMAIL PROTECTED] wrote: On Mon, 30 Jan 2006, Dmitry Ivanov wrote: I have created dynamic CGI-like TFTP server so I will create config files on-the-fly. Now we use this system (dynamic tftp server and Perl CGI script) for country-wide Sipura 3000 configuration. BTW, if anyone is interested I can send sources of this TFTP server. you know you can provision sipura 3000 via http, right? Yes, and I did it before TFTP. But some other equipment requires TFTP, and we decided to use single server. -- The PSTN will never be a slave to you. You must be a slave to it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Most Popular FREE SoftPhone for Windows
Dave Morrow a écrit : Hi all. I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX? If you have the choice, go with IAX. I'm using IaxComm and Diax. They work great, Diax is multi language, IaxComm works Windows and Linux, no FW issues, etc. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guenther Boelter Look here for the updated firmware: http://www.grandstream.com/BETATEST/ Don't ask me why, but you really have to use capital-letters for the word BETATEST!! If you are interested in 1.0.7.11beta, i can gsend you a copy via email because it's not on the server anymore. Thanks. I upgraded just one phone. I noticed new options for provisioning and upgrading: any help abouth these? I'm trying to setup a central HTTP server with a dir for firmware files (and it is OK) and another for configuration files. I read that phone looks for config files (cfg.txt and cfgmac-address) at startup but I'm seeing only one request: GET /grandstream/config/cfg000b82081c55 I'd like to have a general cfg.txt file for all phones and a specific cfgmac-address for any phone. Am I wrong? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
Can anyone explain me differences among: - chan_capi (and chan_capi-cm) - bristuff - mISDN ? Thanks Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, January 31, 2006 11:12 AM we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New GXP-2000 Beta firmware available
From the usual place, http://www.grandstream.com/BETATEST/GXP2000/
Note, there are two (and it took me a bit of a while to figure out)
images to be loaded. Copy the ...a.bin's and the .bin's to your http
provisioning directory, and reboot. The phone _must_ load the .bin
files before it understands the ..a.bin files.
After it loads the first one, the phone does lock up with the
'Grandstream' logo displayed. I left it sitting there for a minute just
to make sure it wasn't flashing itself or anything, then power cycled
it. Then it requested the a.bin files, and away it went.
For those that didn't see the unofficial beta firmware, these phones now
support BLF and call pickup (but, in a very asterisk-centric way). Use
your standard HINT to get the BLF, but when the user pushes the flashing
button, the phone sends '**[xtn]'. So you want something in your
dialplan like:
Exten = _**XXX,1,Pickup(${EXTEN:2})
--Rob
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[Asterisk-Users] Asterisk hardware.
Hello all, Just a question, on asterisk box : I looking on the web , for asterisk at large , and 'asterisk future of telephonie' ... If we would like to change our OLD PABX 600 phone with 4 E1, to install a asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with voicemail, zap channels and some agi script ? thanks Fabrice ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] information on how to use asterisk for telephony boards other than given ones
Hi All, I would be happy if anyone can tell me how does asterisk interact with the telephony boards.what files or APIs are used by it to interact with them. thanks and regards krishna ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme and dtmf
Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos conference room. At this point iax client joins to the conference and talking to the first zap channel need to send dtmf to the IVR. Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. Is there someone that can help me? any suggestion i welcome. Best Regards Accursio Avona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Preventing Asterisk from transfering the call
A user has set in his phone to transfer each call to another number. Is it possible to configure Asterisk not to transfer the calls? Or is it only phone setting? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On Tue, 31 Jan 2006, Mimmus wrote: Can anyone explain me differences among: - chan_capi (and chan_capi-cm) If your card and its driver support a CAPI 2.0 interface, you should use chan_capi-cm. Eicon DIVA Server, AVM and some other which I don't know. - bristuff I'm not the expert here, but AFAIK this is for HFC based cards only. - mISDN This is for almost all passive cards (formaly HiSax driver). But there are others who know more here. Armin ? Thanks Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, January 31, 2006 11:12 AM we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardware.
Fabrice a écrit : Hello all, Just a question, on asterisk box : I looking on the web , for asterisk at large , and 'asterisk future of telephonie' ... If we would like to change our OLD PABX 600 phone with 4 E1, to install a asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with voicemail, zap channels and some agi script ? Short answer: yes. Long answer. If I had to do something like this, I would: 1) Buy a big box (the one I just bought is a dell 1850 with redundant power supply, raid1 disks, etc) - see dell.com 2) Grab a digium card: http://www.voipsupply.com/product_info.php?products_id=913 3) Buy some decent SIP phones: http://www.voipsupply.com/product_info.php?products_id=758 (I haven't tried those yet but they have a good reputation) As for the Asterisk distro I really like Xorcom Rapid: http://www.xorcom.com/ It's debian based, it's clean, and I'm sure the author won't mind you hiring him to get the 4E1 card working and configure the distro to your liking. Once it's done, save the disk image (using dd, or mindi / mondo) to make sure you can redeploy quickly in case of emergency... All in all you're probably looking at €15-20k for a typical PBX replacement. You probably want to look into VoIP as well to reduce operational costs... Old PSTN providers are expensive, with some IP routing (at least for some of the outbound calls) you can probably recoup the investment over a few years thanks to phone savings. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Web interface
On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote: +++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. And unlike AMP, it is non-free. BTW: there is also DeStar: http://destar.berlios.de/ . Version 0.1.1 was recently released. Nice and clean. Generally runs its own daemon, though can run under apache. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Individual SIP account how to make it Trunk
Hi, i have diffirent provider example(3 single account in deltathree, 4 account in packet8 and so on) . How this possible to make the three individual sip account in deltathree act as trunk so that i cannot get a busy call. If line one fail goto line 2 then line 3 or another trunk line 1 then line 2 then line3I read it in asterisk at home but the script i am copying is not working . any help is very much appreciated.. TIA //jollyr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
Can someone tell me the advantage in using an active card such as the AVM-B1 do they have echo cancelling built in? Just that I've got three pots lines and keep thinking I should convert over to ISDN but I don't want to get echo issues. Chris - Original Message - From: Armin Schindler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 31, 2006 11:35 AM Subject: RE: [Asterisk-Users] Interface card for Euro-ISDN (BRI) On Tue, 31 Jan 2006, Mimmus wrote: Can anyone explain me differences among: - chan_capi (and chan_capi-cm) If your card and its driver support a CAPI 2.0 interface, you should use chan_capi-cm. Eicon DIVA Server, AVM and some other which I don't know. - bristuff I'm not the expert here, but AFAIK this is for HFC based cards only. - mISDN This is for almost all passive cards (formaly HiSax driver). But there are others who know more here. Armin ? Thanks Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, January 31, 2006 11:12 AM we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Individual SIP account how to make it Trunk
Jolly M. Recto wrote:
Hi,
i have diffirent provider example(3 single account in deltathree, 4
account in packet8 and so on) . How this possible to make the three
individual sip account in deltathree act as trunk so that i cannot get
a busy call. If line one fail goto line 2 then line 3 or another trunk
line 1 then line 2 then line3I read it in asterisk at home but the
script i am copying is not working .
Have you had a look at GetGroupMatchCount?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GetGroupMatchCount
and
from page http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup is
the example below:
Example 7 Using Categories in 1.2.x (Ramon's example)
Using categories you are able to set multiple groups on only one active
channel.
So you are able to set the amount of calls on the called channel but
also on the calling channel.
exten = 200,1,Set(GROUP(${EXTEN})=OUTBOUND_GROUP)
; Increase number of calls on the called channel
exten = 200,n,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
1]?BLOCK)
; Check if called channel has more than 1 call.
exten = 200,n,Set(GROUP(${CALLERIDNUM})=OUNTBOUND_GROUP)
; Increase number of calls on the calling channel
exten = 200,n,Dial(SIP/200)
; Call the extension
exten = 200,n(BLOCK),Busy
; GotoIf jumped here if the was more than 1 call using labels
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[Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]'(case =0)". my configuration files are attached below. any help would be greatly appreciated. many thanks in advance.ABC abc def [EMAIL PROTECTED] wrote:there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls).thanks in advance for any hint or suggestion. AmaI just post my configuration file here for sip phone:
extensions.conf-[globals] [default]include = incominginclude = outgoinginclude = iaxinculde = sipinclude = sccp[sip]exten = 2171,1,Dial(SIP/stargate1,20);exten = 2171,1,Dial(SIP/2171,20)exten = 2171,2,Hangupexten = 2172,1,Dial(SIP/stargate2,20);exten = 2172,1,Dial(SIP/2172,20)exten = 2172,2,Hangupexten = 2173,1,Dial(SIP/stargate3,20);exten = 2173,1,Dial(SIP/2173,20)exten = 2173,2,Hangup [sccp] [skinny] [incoming]exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)exten = _214943[5-9]6,2,Hangup [outgoing]exten = _,1,Dial(Zap/g1/${EXTEN})exten = _,2,Hangup-
sip.conf-[general]context=default ; Default context for incoming calls ; Set this to your host name or domain namebindport=5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup=yes ; Enable DNS SRV lookups on outbound
calls register = stargate1:[EMAIL PROTECTED]/2171register = stargate2:[EMAIL PROTECTED]/2172register = stargate3:[EMAIL PROTECTED]/2173;-- NAT SUPPORT nat=no ; Global NAT settings (Affects all peers and users) [local_sip]type=friendhost=10.47.200.136context=default [stargate1] ;cisco
9760;[2171]type=friendhost=dynamic ;10.47.200.140 ;dynamicdefaultip=10.47.200.140username=stargate1secret=xxxcallerid="21495071" 2171allow=allqualify=200nat=nodefaultip=10.47.200.140 [stargate2] ;Polycom 601;[2172]type=friendhost=dynamic ;10.47.200.141 ;dynamicdefaultip=10.47.200.141username=xxxsecret=2stargatecallerid="21495072" 2172allow=allqualify=200nat=nodefaultip=10.47.200.141 [stargate3] ;Aastra 480i;[2173]type=friendhost=dynamic ;10.47.200.137 ;dynamicdefaultip=10.47.200.137username=stargate3callerid="starg ate3" 2173secret=xxxallow=allqualify=200nat=nodefaultip=10.47.200.137 [EMAIL PROTECTED] wrote: What error do you get when trying to call the SIP phones?PaulH - Original Message - From: abc def To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 11:58 PM Subject: [Asterisk-Users] Help with sip setup because can't receive calls Hi all, I readmany posts on asterisk mail site and been trying many different thingsbut still I can't get my sip phones to work with asterisk. I have a full blown-up voip netwok with two asterisk servers connected to pstn networkwith iax phones and cisco sccp phones which all work fine. however, I have been struggeling to configure my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip phones to anywhere else but not receive phone calls. I can see the phones on "sip show registry" and "sip show peers" but no track phone calls for sip. can you please shed some light on me how to go about solving this problem? thank you and best regards, Ama<
HR
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Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On 1/31/06, John Jensen [EMAIL PROTECTED] wrote: Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Only have experience with junghanns cards, but they are the same.. beronet doesn't use bristuff.. but you can also use junghanns cards the beronet-way.. have a look on voip-info.org, some usefull info on these BRI cards can be found there. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Kirk IP600
On 1/31/06, Giordano Grandis [EMAIL PROTECTED] wrote: [chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module chan_sccp.so failed! check the chan_sccp homepage, make sure you 'clean up' your asterisk modules and include directories.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue() with timeout=0
Hello,
i've recently switched over from 1.0.9 to 1.2.3.
I've experienced some (to me) weird behaviour.
This is the config for an example queue.conf:
[654]
wrapuptime=30
timeout=20
strategy=ringall
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
music=default
monitor-join=yes
monitor-format=
maxlen=0
leavewhenempty=no
joinempty=no
context=
announce-holdtime=no
announce-frequency=45
extensions.conf
exten = 654,1,Answer
exten = 654,2,SetCIDName(${CALLERIDNAME})
exten =
654,3,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMEST
AMP}-${UNIQUEID})
exten = 654,4,Queue(654|t|||0)
exten = 654,5,Goto(ext-queues,654,1)
now when I place a call into the queue the agent times out after 20secs and
the dialplan executes the next
step instead of keeping the call into the queue for an unlimited time which
I thought a nul-value as the
timeout variable would do (Queue(654|t|||0)).
Could anyone tell me if I can still use zero (0) as a value for unlimited in
the command Queue(654|t|||0).
thanks,
Bart
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Re: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
Hi how about SIP friend to SIP Friend even it taking gsm ram On 1/31/06, JP Carballo [EMAIL PROTECTED] wrote: ram wrote: Hi as per the list people guidence i have downloaded the Codec and installe my Pc is P4, but i have downloaded the P2.so file and copied in specific directory whe i see show translation i could able to see 30 i have configure AAH for VOIP JET connection when i try to make call out, its using only GSM even though i mention g729 in top list whats wrong ?? ramIMHO voipjet only allows g.711 ulaw. --JP Carballohttp://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Kirk IP600
Giordano Grandis ha scritto: I installed the chan_sccp and configured the sccp.conf, but when try to start asterisk I get this error [chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module chan_sccp.so failed! you have to load the module res_features.so Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
John Jensen schrieb: I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. How many lines do you want to terminate? Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with outbound call but there is no
inbound call to my sip phone. the only message I get when I call from pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def [EMAIL PROTECTED] wrote:
there is no error message coming up on the pbx for in-bound calls (there is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT SUPPORT
nat=no ; Global NAT settings (Affects all peers and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls
Hi all,
I read many posts on asterisk mail site and been trying many different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601,
Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
phones to anywhere else but not receive phone calls. I can see the phones on
sip show registry and sip show peers but no track phone calls for sip.
can you please shed some light on me how to go about solving this
problem?
thank you and best regards,
Ama
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Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with outbound call but there is no
inbound call to my sip phone. the only message I get when I call from pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def [EMAIL PROTECTED] wrote:
there is no error message coming up on the pbx for in-bound calls (there is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT SUPPORT
nat=no ; Global NAT settings (Affects all peers and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls
Hi all,
I read many posts on asterisk mail site and been trying many different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601,
Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
phones to anywhere else but not receive phone calls. I can see the phones on
sip show registry and sip show peers but no track phone calls for sip.
can you please shed some light on me how to go about solving this
problem?
thank you and best regards,
Ama
HR SIZE=1 Do you Yahoo!?
With a free 1 GB, there's more in store with Yahoo! Mail.
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Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On Tue, 31 Jan 2006, Chris Stenton wrote: Can someone tell me the advantage in using an active card such as the AVM-B1 do they have echo cancelling built in? Just that I've got three pots lines and keep thinking I should convert over to ISDN but I don't want to get echo issues. The active cards do the ISDN protocol stuff on board, so the host CPU/driver does not need to do that - better performance, less interrupts. The AVM cards do not have such DSPs on board, so no echo-cancel. But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, DTMF-detection, voice codec, line interconnect/mixing even to other cards, ... with the on boards DSP. chan_capi-cm is supporting most of the features and will do more soon. Armin Chris - Original Message - From: Armin Schindler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 31, 2006 11:35 AM Subject: RE: [Asterisk-Users] Interface card for Euro-ISDN (BRI) On Tue, 31 Jan 2006, Mimmus wrote: Can anyone explain me differences among: - chan_capi (and chan_capi-cm) If your card and its driver support a CAPI 2.0 interface, you should use chan_capi-cm. Eicon DIVA Server, AVM and some other which I don't know. - bristuff I'm not the expert here, but AFAIK this is for HFC based cards only. - mISDN This is for almost all passive cards (formaly HiSax driver). But there are others who know more here. Armin ? Thanks Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, January 31, 2006 11:12 AM we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. How many lines do you want to terminate? Two to Four ISDN2 lines. That gives me a maximum of eight voice channels. /John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P FXO port problem.
I have a digium TDM400P with 1 FXO and 1 FXS port. I have a standard analog phone connected to the FXO port and place calls to the PTSN phone line. The analog trunk is accessed via the standard 9 and area code (if needed) and of course the phone number. The error is as follows. I dial 9,866-XXX- some other number is dialed such. This occurs with toll free as well as local and or long distance numbers. My SIP phones work well as do any softphones. I know this must be a cockpit problem so any assistance grealy appreciated. I assume the above is a typo; analog phones connect to the FXS port, not FXO. (The analog phone wouldn't function at all if it was plugged into a FXO port.) There really isn't anyway to guess at your dialing problem without you providing copy/paste portions of your configs. It sort of sounds like your analog phone is dropping into an extensions.conf context that is different from what your sip phones are using. Paste the appropriate sections of zapata.conf and extensions.conf so we can see what you're doing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie dial problem,
Hello friends, I am using asterisk with sip phones and sip fxo box. My problem is that my dtmf is recognised internally only if I use dtmf=inband and outside to the pstn lines work only if I use dtmf=info. The result is that I cant transfer any calls from and to pstn. How do I fix this. Either one works properly or the other but not both of them. So when I have configured my boxes as dtmf=inband and I dial them inhouse. When I have to make a call outside, I say SipDtmfMode=info and dial outside. But then it doesnot transfer the call. Please help me. I am stuck up. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipbuster incoming
Hi all, Some friends of mine have an asterisk box which they use for outgoing IAX2 via voipbuster.com. They have been told that they now have an incoming number 0044117*** The thing is I cant seem to get any debug info on the incoming. I have tried both sip and IAX trunks but dont see any incoming info. Anyone have any idea what protocol voipbuster use for incoming calls?? Thanks in advance Bails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to start a playback after the called party picks up?
1. I want to call somebody and, as soon (and not before) a playback should be played. How can I do that? 2. How can I accept dtmf tones with such calls? Example: System calls all staff and ask them a question. The staff will answer with a digit! The playback should start when the staff picks up. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301: Pass-through ethernet port unusable?
What version of the firmware? Jerry Glomph Black wrote: Have just done a deployment of 45 of these puppies. They are doing their main job quite well, but of course there are minor kinks. A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack, as soon as you send any reasonable amount of traffic (even casual web surfing) the phone seizes. We had to run a bunch of cables in a big rush to users' PCs, having (erroneously) believed that the passthru RJ45 would be a usable port! Has anyone out there experienced this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gain adjustment
Hi! When adjusting the rxgainand txgain inAsterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? -- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom ip601 attendant console
Using the Buddy Watch functionality on the IP601 you can watch up to 6 people. The expansion modules are not good for much more than speed dials due to this limitation. After talking to our vendor, the reason it is limited to 6 is due to the current version of Asterisk Business Edition's lack of [documented/advertised] support for SUBSCRIBE/NOTIFY. Since Polycom is certified against ABE and the most recent release of ABE doesn't support this functionality, Polycom will not open their firmware up to allow more than six. I am hoping someone from Digium is monitoring this thread and that they might comment on when the new edition of ABE will be released so that we can actually utilize the full capabilities of the IP601's attendant consoles. Right now they (the attendant consoles) are pretty useless to me. Has anyone else had any success with them? Saul Diaz wrote: Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It works beautifull and not issues. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom ip601 attendant console
Just curious, I have had issues with the number of monitored phones and getting out of sync with reloads. Have you had similar issues? Which version of *? On Jan 30, 2006, at 7:04 PM, Saul Diaz wrote: Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It works beautifull and not issues. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
ram wrote: Hi how about SIP friend to SIP Friend even it taking gsm ram Check the [general] section of your sip.conf Most likely there is an allow=gsm line there. Just allow=ulaw on your end so you can connect to voipjet. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER redirect
my setup is client--registers-- ser-redirect---client ---invite-- asterisk -- pstn when this happens i configured the ser.cfg with the rewriteuri and redirect logic and i am seeing 300 redirect being passed to the client registerd to ser but when it sends a invite to asterisk, asterisk looks for the same ip address of the client to send reply to and i receive a error on the asterisk server realtime_peer: Cannot Determine peer name ip=xxx.xxx.xxx.xxx I would appreciate if someone can help me figure this out. Thank you, AAOn 1/30/06, Velimir Novkovic [EMAIL PROTECTED] wrote: Check http://www.voip-info.org/wiki/view/Asterisk+at+largeOr sipedu http://mit.edu/sip/sip.edu/Plenty of examples /Vel-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of SharonSent: Friday, January 27, 2006 4:41 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] SER redirecthello, can someone help me with ser redirect to asterisk.any help appreciated.Thanks,AA___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI) [ Virusgeprüft]
with incoming lines only maybe are active capi dual/quad-port cards from AVM an alternative - but I've no experience with them together with asterisk/chan_capi an other way with 4 isdn-lines is to think about to order an partial E1 line with 8 channels... [EMAIL PROTECTED] wrote on 31.01.2006 14:17:59: I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. How many lines do you want to terminate? Two to Four ISDN2 lines. That gives me a maximum of eight voice channels. /John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain adjustment
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP301: Pass-through ethernet port unusable?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jerry Glomph Black Sent: Monday, January 30, 2006 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port unusable? Have just done a deployment of 45 of these puppies. They are doing their main job quite well, but of course there are minor kinks. A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack, as soon as you send any reasonable amount of traffic (even casual web surfing) the phone seizes. We had to run a bunch of cables in a big rush to users' PCs, having (erroneously) believed that the passthru RJ45 would be a usable port! Has anyone out there experienced this? No issues on the IP501 with 2.6.2 bootrom and 1.5.3 SIP. Ethernet port works fine for the PC. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
Hi yes i have added all of them in allow one by one like this allow=g729 allow=gsm allow=ulaw allow=alaw ram On 1/31/06, JP Carballo [EMAIL PROTECTED] wrote: ram wrote: Hi how about SIP friend to SIP Friend even it taking gsm ramCheck the [general] section of your sip.confMost likely there is an allow=gsm line there.Just allow=ulaw on your end so you can connect to voipjet.--JP Carballo http://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing 2 channels at the sametimewithdifferentcaller ID number?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Tuesday, January 31, 2006 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels at the
sametimewithdifferentcaller ID number?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Monday, January 30, 2006 11:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels at the same
timewithdifferentcaller ID number?
Exten = _NXXNXX,2,Set(__ORIGCID=CALLERID(number))
exten =
_NXXNXX,2,dial(sip/${EXTEN}local/[EMAIL PROTECTED]/n,r}
[alternate1]
exten = _NXXNXX,1,macro(alternate-number|${__ORIGCID})
[macro-alternate-number]
exten = s,1,set(CALLERID(number)=${ARG1}) exten =
s,2,dial(SIP/[EMAIL PROTECTED])
This sets _ORIGCID = CALLERID(number), I think you meant
Set(_ORIGCID=${EXTEN})
??
I will give it a try.
-- Accepting call from '3035551212' to '3035551313' on
channel 0/19, span 1
THIS exten = 3037687402,1,set(CALLEDNUM=${EXTEN})
RESULTS IN THIS -- Executing Set(Zap/19-1,
CALLEDNUM=3035551313) in new stack
OK SO FAR, VARIABLE SET
-- Executing Dial(Zap/19-1,
local/[EMAIL PROTECTED]/n) in new stack
-- Called [EMAIL PROTECTED]/n
THIS exten = _NXXNXX,1,macro(alternate-number|${CALLEDNUM})
RESULTS IN THIS -- Executing
Macro(Local/[EMAIL PROTECTED],2,
alternate-number|) in new stack
PROBLEM! ${CALLEDNUM} is no longer = 3035551313
Channel variable will not pass from original macro to local channel...
Did I miss something?
Yes you must prefix a variabel with __ that's (2) _ underscores so that
it cross channels.
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Re: [Asterisk-Users] Gain adjustment
On 1/31/06, Rich Adamson [EMAIL PROTECTED] wrote: When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk oris it enough to just reload Asterisk in order to apply the new setting?Need to stop asterisk and restart it. A reload will not take the new settinginto consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. OK. If I set the gain to a negative number then i decrease the volume? And a positive number increases the volume? -- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing 2 channels at thesametimewithdifferentcaller ID number?
Yes you must prefix a variabel with __ that's (2) _ underscores so that it cross channels. Aah, the magic formula - documented where? :) Thanks a million, have a great day. Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 1 FXO Problem
I have Asterisk 1.2 and a generic Wildcard single FXO card (a cheapo from eBay). I have read about many people who have used these cards without an issue and I'm just testing to work up a new system. The problem I have is that if I call the telephone number of the line attached to that card and have configured my incoming calls to, say, forward to an extension, then it will ring that extension as expected but if you pick up that extension then the extension just hears dial tone and the originator never hears that anyone picked up, it just keeps ringing. On the other hand if you set it to go to a VM or auto attendant, the system never picks up. I have run in debug mode to see the messages and it appears to be going through my config just fine, I'll see ANSWER but like I said it never picks up. Any ideas what would cause this? I just bought a Digium TDM02B in hopes that this might help, but I would like to be able to use the 1FXO cards as well. Thank You! ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain adjustment
On Tue, Jan 31, 2006 at 08:44:18AM -0600, Rich Adamson wrote: When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. Actually there is a restart in chan_zap of Asterisk 1.2. And zapata.conf is of chan_zap. However I'm not sure exactly how much of those changes do apply on repload. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP501 Endless Loop
I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers. Anybody have an idea as to how to get around this? I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). Thanks! ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing 2 channels atthesametimewithdifferentcaller ID number?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, January 31, 2006 8:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels
atthesametimewithdifferentcaller ID number?
Yes you must prefix a variabel with __ that's (2) _ underscores so
that
it cross channels.
Aah, the magic formula - documented where? :)
Thanks a million, have a great day.
Damon
Man, if I could only learn to read...
I looked at this page 5 times and never saw the plain as day answer to
my question.
From the wiki;
Inheritance of Channel Variables
Prepending a single _ character to a variables name in SetVar will cause
that variable to be inherited by channels created by the main channel.
eg. when using Dial(Local/...); once inherited these variables will not
be further inherited. Prepending two _ characters will cause them to be
inherited indefinitelty. (Only works in CVS HEAD, not yet implemented in
Asterisk 1.0.9.)
Note that for retrieval purposes these variable names do not need to
include the underscores.
[TestInherit]
exten = 100,1,SetVar(__FOO=5)
exten = 100,2,Dial(Local/[EMAIL PROTECTED])
exten = test,1,NoOp(${FOO})
will result in FOO being inherited. Without the underscores, the new
local channel would start with a clean slate.
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Re: [Asterisk-Users] polycom ip601 attendant console
These were configured with the MACADDR-directory.xml and the 6 extension limitation has been verified by several vendors. Don't get me wrong, they are a nice looking unit, and once the monitoring of more than 6 people is available they will be a great replacement for the Snoms. Right now we can transfer to any of the speed dials on those buttons but those buttons do not show the state of those extensions when transferring. To clarify, you can have as many speeddials/extensions assigned as you have buttons on the expansion modules, but you can only monitor or watch six. Using OEJ's parkhints patch (bug 5779) you can actually monitor the state of parking spots too, so once this limitation of 6 watched extensions is changed we'll have a really sweet solution for a receptionist phone. Damon Estep wrote: Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It works beautifull and not issues. regards Saul Saul, Perhaps you could tell us a little more about what you have it set up to do. Can you monitor 20 extensions (6 on the phone, 14 on the console)? What about transferring a call to a monitored extension using the console button? What does your config look like? Another user posted that the IP 601 is limited to monitoring 6 extensions, but I wonder if that is just via the web interface and the config file method of setup allows more? Thanks. I have not purchased one of these yet and wanted to see if anyone has had success before buying a brick for $200+ Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preventing Asterisk from transfering the call
well. Im supposing you mean a SIP phone. Transfers with SIP phones happens to be a method called REFERRER. Im not sure if its a feature of Asterisk to allow the administrator to ban the referrers, but if is not a feature, letme know, may be i can make a patch soon. To look for a feature like that, check voip-info.org, in the sip.conf section regardsOn 1/31/06, Bartosz Piec [EMAIL PROTECTED] wrote: A user has set in his phone to transfer each call to another number. Isit possible to configure Asterisk not to transfer the calls? Or is itonly phone setting?--Best regards,Bartosz Piec___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help configuring Asterisk server
please consider posting this as a Job offer in asteriskhelpdesk, because of your lack of information i can tell you are really stuck :DOn 1/30/06, Naren Koka [EMAIL PROTECTED] wrote:I need to configure / migrate Asterisk server from 0.9 to the latest version with some upgrades. Please help!Thank you.Sincerely,Naren Koka(480) 829-0479___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain adjustment
Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. OK. If I set the gain to a negative number then i decrease the volume? And a positive number increases the volume? That's correct. Don't bother with 1/10ths. Just use -2 or 4 or 0 or whatever. You can enter tenths, but its really not going to get you anywhere. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain adjustment
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. Actually there is a restart in chan_zap of Asterisk 1.2. And zapata.conf is of chan_zap. However I'm not sure exactly how much of those changes do apply on repload. Unless someone just change this recently, the gain settings are not changed on a reload. Since I've been watching the svn/cvs updates rather closely, I don't believe any changes have actually been made (but I could have missed them as well). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to register using SIP
Sorry for the duplicate post but I have hit a brick wall trying to get this to work. Is there anyone who can help me? I am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing 2 channelsatthesametimewithdifferentcaller ID number?
Don't feal bad about not reading. I yell at my 10 y.o. about it all the
time. READ, NO more TV, READ!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Tuesday, January 31, 2006 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2
channelsatthesametimewithdifferentcaller ID number?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, January 31, 2006 8:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels
atthesametimewithdifferentcaller ID number?
Yes you must prefix a variabel with __ that's (2) _ underscores so
that
it cross channels.
Aah, the magic formula - documented where? :)
Thanks a million, have a great day.
Damon
Man, if I could only learn to read...
I looked at this page 5 times and never saw the plain as
day answer to my question.
From the wiki;
Inheritance of Channel Variables
Prepending a single _ character to a variables name in SetVar
will cause that variable to be inherited by channels created
by the main channel.
eg. when using Dial(Local/...); once inherited these
variables will not be further inherited. Prepending two _
characters will cause them to be inherited indefinitelty.
(Only works in CVS HEAD, not yet implemented in Asterisk 1.0.9.)
Note that for retrieval purposes these variable names do not
need to include the underscores.
[TestInherit]
exten = 100,1,SetVar(__FOO=5)
exten = 100,2,Dial(Local/[EMAIL PROTECTED])
exten = test,1,NoOp(${FOO})
will result in FOO being inherited. Without the underscores,
the new local channel would start with a clean slate.
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[Asterisk-Users] Forwarding issue.
If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI -- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.20.1.79:5060;branch=z9hG4bK271f9f8f From: asterisk sip:[EMAIL PROTECTED];tag=as319b6dd5 To: sip:[EMAIL PROTECTED];tag=A368FBB2-B6EAC0CF CSeq: 104 BYE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 Aside from the MoH and the error, all procedes as it should; eventually, the call is either answered or goes to voicemail. Any ideas as to what I'm doing wrong? Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 Endless Loop
Sounds like the phone cannot log into the FTP server. Did you create the proper user with the correct login? It's set up in the FTP/TFTP menu.Also, you can end the loop by just going into the config menu and nuking the FTP info and then you'll get a message that says it could not contact the boot server. On 1/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots.It is annoying like nothing I can describe!I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers.Anybody have an idea as to how to get around this?I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). Thanks!___Sent by ePrompter, the premier email notification software.Free download at http://www.ePrompter.com .___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 Endless Loop
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] said: I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers. Anybody have an idea as to how to get around this? I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). Since you are running Asterisk, it would make sense to use a Linux based FTP server. At least then you would have decent logging (turn on verbose logging) which you can post the output of. I would also suggest sniffing the FTP attempt with ethereal or tcpdump to get more info on it. In any case, you are going to have to get more details: When you say session, is it actually logging in correctly? Finding the files it is looking for? Or is it just a connection attempt? My guess is that it either is not logging in correctly or is not finding the files it wants, or it IS finding a file but doesn't like it. Possibly one or more of the files is corrupt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER redirect
Sharon a écrit :
my setup is
client--registers-- ser-redirect---client ---invite--
asterisk -- pstn
when this happens
i configured the ser.cfg with the rewriteuri and redirect logic and i
am seeing 300 redirect being passed to the client registerd to ser but
when it sends a invite to asterisk, asterisk looks for the same ip
address of the client to send reply to and i receive a error on the
asterisk server
realtime_peer: Cannot Determine peer name ip=xxx.xxx.xxx.xxx
I would appreciate if someone can help me figure this out.
I have the same setup and here is what I do:
In ser.cfg:
# -
# Pass on stuff going to PSTN to Asterisk
# -
if (uri=~^sip:[EMAIL PROTECTED]) {
rewritehostport (*your_asterisk_box_ip*:5060);
if (!t_relay()) {
# sl_send_reply (403, prout);
sl_reply_error();
};
break;
};
In sip.conf: (asterisk)
[ser-stuff]
type=friend
context=world
host=my_ser_host
canreinvite=no
Also be careful. If someuser@yourserbox.ip calls not to have any
[someuser] sections in sip.conf, because it broke stuff for me.
Good luck!
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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Re: [Asterisk-Users] cdrtool
i understand.. anyone know how much is basic support from them ? On 1/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello,Call sip:[EMAIL PROTECTED] Regardsharry--- Jimmy Smith [EMAIL PROTECTED] a écrit : anyone having weird problems on latest cdrtool? #!/usr/bin/php4 *Fatal error*: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in */var/www/CDRTool/SOAP/client_lib.php*on line *2 always get weird error like that * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.Téléchargez sur http://fr.messenger.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] international caller id on UK (BT) PRI
When a call arrives on our PRI from a UK domestic number, the presented caller ID looks something like 1223123456. In my dialplan, I stick 90 on the front in order to turn this into a valid number for outward dialling, and everything works fine. However, when a call comes in from an international number, I need to add an extra zero -- that is, 491234123456 needs to have 900 added on the front to make it valid. Is there some Asterisk variable I can inspect to find out whether the presented CLI is using a national or international number plan? thanks p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.1 + TDM400P + fax machine unreliable ?
Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine works very reliable with a standard Telco POTS line. Who has a 'good' working solution to connect an ordinary Fax Machine to Asterisk ? Thanks for any help. Alex Parts of my setup: Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1d built by root @ pbx on a i686 running Linux on 2006-01-05 13:25:25 UTC # cat /etc/zaptel.conf loadzone=be defaultzone=be span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 span=3,1,3,ccs,ami bchan=7-8 dchan=9 span=4,1,3,ccs,ami bchan=10-11 dchan= 12 # TDM400P fxoks=13-16 # cat /proc/zaptel/5 Span 5: WCTDM/0 Wildcard TDM400P REV I Board 1 13 WCTDM/0/0 FXOKS (In use) 14 WCTDM/0/1 FXOKS (In use) 15 WCTDM/0/2 FXOKS (In use) 16 WCTDM/0/3 FXOKS (In use) Part of /etc/asterisk/zapata.conf: [channels] ; Default language language=be callerid=asreceived immediate=no switchtype=euroisdn cut the ISDN (bristuff) part of it.. ; ; TDM40B kanalen ; signalling=fxo_ks language=be context=analog echocancel=no channel = 13 signalling=fxo_ks language=be context=analog echocancel=no channel = 14 signalling=fxo_ks language=be context=analog echocancel=no channel = 15 signalling=fxo_ks language=be context=analog echocancel=no channel = 16 # Part of extensions.conf ; filter de Fax exten = 15504409,1,Dial(Zap/13,60) exten = 15504409,2,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdrtool
Ok anyone have latest cdrtool running 4.1 i think.. ill pay for install On 1/31/06, Jimmy Smith [EMAIL PROTECTED] wrote: i understand.. anyone know how much is basic support from them ? On 1/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello,Call sip:[EMAIL PROTECTED] Regardsharry--- Jimmy Smith [EMAIL PROTECTED] a écrit : anyone having weird problems on latest cdrtool? #!/usr/bin/php4 *Fatal error*: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in */var/www/CDRTool/SOAP/client_lib.php*on line *2 always get weird error like that * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.Téléchargez sur http://fr.messenger.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail greetings
Hi, I`ve been trying to figure out voicemail, but there is something that is obviously escaping me. Using * 1.2.3, standard built with asterisk-addons. I have two voicemails, one is 702 and one is 705. Both in different contexts, but that doesn`t matter (I think). The point is in the /voicemail/context/702 directory I have the files unavail.gsm, temp.gsm and greet.gsm. While in the other directory, I have greet.gsm, unavail.gsm and busy.gsm. So in one directory I have temp.gsm and in the other busy.gsm. How did that happen and what does it mean? What i found out is that in the one voicemail that doesn`t have temp.gsm, when somebody tries to leave me a message that person gets an asterisk greeting (as opposed to one with my wonderful voice). Also, WHEN are the file used? I have the option of recording my busy message and my unavailable message, but really, how does Asterisk choose which one I am? (unavailable vs busy)? This isn`t clear to me, hopefully somebody has a quick and simple answer. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to start a playback after the called partypicks up?
Ronald, I've been experimenting with something similar. You might want to check this out: http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message What kind of trunks do you have for your outbound calls? (BRI/PRI/analog POTS/SIP/IAX etc.) I'm using PRI and it works very well - the dialplan doesn't execute the message playback until after the call has been answered. I don't know of any analog lines that can do that. (However, there is app_amd.c that tries to detect an answering machine vs. human answer which might suffice for dialing on analog lines.) BTW, the wiki has some nice dialplan examples that I pasted right into my extensions.conf file and I was working in no time. Please let us know if this helps. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Tuesday, January 31, 2006 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to start a playback after the called partypicks up? 1. I want to call somebody and, as soon (and not before) a playback should be played. How can I do that? 2. How can I accept dtmf tones with such calls? Example: System calls all staff and ask them a question. The staff will answer with a digit! The playback should start when the staff picks up. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gain adjustment
FYI I just tested on * 1.2.1 a reload chan_zap.so It takes the new settings from zapata.conf. I know because I changed the context and after a reload it showed the new context. I can only assume that the gain settings are also changed. Regards Allan Gee Phone: +27 21 4644400 Ext. 103 www.equation.co.za -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tzafrir Cohen Sent: 31 January 2006 05:17 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Gain adjustment On Tue, Jan 31, 2006 at 08:44:18AM -0600, Rich Adamson wrote: When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. Actually there is a restart in chan_zap of Asterisk 1.2. And zapata.conf is of chan_zap. However I'm not sure exactly how much of those changes do apply on repload. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER redirect
i have ser and asterisk on 2 different boxes.
my ser.cfg
if (method==REGISTER) {
if(!www_authorize(ser domain name, subscriber)){
www_challenge(ser domain name, 0);
break;
}
sl_send_reply(200, ok);
break;
};
rewritehostport (ip addr of asterisk box:5060);
sl_send_reply (300, redirect);
}
asterisk setting in sip.conf:
i am not adding ser as a peer neither am i adding the peer registered with ser in the sip.conf
i wanted ser to pass a redirect to the client registered with ser (this part works)
then ser is out of the call and the client and asterisk talk but on my asterisk box i'm seeing the following error
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to xxx.xxx.xx.xx : 5060 (NAT)
chan_sip.c:realtime_peer: Cannot Determine peer name ip=xx.xxx.xxx.xxx
Found no matching peer or user for 'xx.xx.xx.xx:5060
its looking for same ip of the ser client to send back the reply.
On 1/31/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Sharon a écrit : my setup is client--registers-- ser-redirect---client ---invite-- asterisk -- pstn when this happens i configured the
ser.cfg with the rewriteuri and redirect logic and i am seeing 300 redirect being passed to the client registerd to ser but when it sends a invite to asterisk, asterisk looks for the same ip address of the client to send reply to and i receive a error on the
asterisk server realtime_peer: Cannot Determine peer name ip=xxx.xxx.xxx.xxx I would appreciate if someone can help me figure this out.I have the same setup and here is what I do:
In ser.cfg:# -# Pass on stuff going to PSTN to Asterisk# -
if (uri=~^sip:[EMAIL PROTECTED]) {rewritehostport (*your_asterisk_box_ip*:5060);if (!t_relay()) {# sl_send_reply (403, prout);sl_reply_error();
};break;};In sip.conf: (asterisk)[ser-stuff]type=friendcontext=worldhost=my_ser_hostcanreinvite=noAlso be careful. If someuser@yourserbox.ip
calls not to have any[someuser] sections in sip.conf, because it broke stuff for me.Good luck!Jean-Michel.--Jean-Michel Hiver - http://ykoz.net/Découvrez la Réunion des Technologies IP Telecom
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RE: [Asterisk-Users] Forwarding issue.
I had this same issue with 601s, and I was able to fix it by defining: progressinband=yes in sip.conf. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Tuesday, January 31, 2006 11:20 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Forwarding issue. If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI -- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.20.1.79:5060;branch=z9hG4bK271f9f8f From: asterisk sip:[EMAIL PROTECTED];tag=as319b6dd5 To: sip:[EMAIL PROTECTED];tag=A368FBB2-B6EAC0CF CSeq: 104 BYE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 Aside from the MoH and the error, all procedes as it should; eventually, the call is either answered or goes to voicemail. Any ideas as to what I'm doing wrong? Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 Endless Loop
I had the /exact/ same problem. Turns out it's the FTP server; in the docs, there are several FTP servers specified as being compatible; proftp is the one I went with, and it fixed it right up. (Note that I was using the default Debian FTP server when it was rebooting, so it's not just a 'doze issue.) -Ken Walt Reed wrote: On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] said: I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers. Anybody have an idea as to how to get around this? I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). Since you are running Asterisk, it would make sense to use a Linux based FTP server. At least then you would have decent logging (turn on verbose logging) which you can post the output of. I would also suggest sniffing the FTP attempt with ethereal or tcpdump to get more info on it. In any case, you are going to have to get more details: When you say session, is it actually logging in correctly? Finding the files it is looking for? Or is it just a connection attempt? My guess is that it either is not logging in correctly or is not finding the files it wants, or it IS finding a file but doesn't like it. Possibly one or more of the files is corrupt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: The active cards do the ISDN protocol stuff on board, so the host CPU/driver does not need to do that - better performance, less interrupts. The AVM cards do not have such DSPs on board, so no echo-cancel. But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, DTMF-detection, voice codec, line interconnect/mixing even to other cards, ... with the on boards DSP. chan_capi-cm is supporting most of the features and will do more soon. Armin While we are at the subject another couple of simple related question. Are HFC-S cards active? I got one for a very low price, so that I imagine it will be NOT the case... What cards do support operation of an ISDN phone set? (I imagine there will be something similar to the FXS-FXO stuff of the analog world in the ISDN land). Thanks in advance Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.1 + TDM400P + fax machine unreliable ?
Alex Ongena wrote: I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400Ptab=support (see the last list item) in other words: it often does not work. regards frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hangs on 1.2.1
Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0', 10 retries! Once this happens, all of my sccp phones drop offline and attempt to register. I get no sccp messages on the console. There's really nothing on the console to indicate any sort of problem. If I try to do an unload chan_sccp.so and then load it back, all of my SIP phones lose their registrations, none of my Zap channels work and I have to kill Asterisk and restart it. Is this an Asterisk problem or an SCCP problem? Help!! Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canadian Termination $0.0039 / Minute
All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: [EMAIL PROTECTED] Thanks, Jon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom ip601 attendant console
To monitor who is doing what we writing a program that every user can have on thier windows desktop to see the status of all phones on the system. It's AIM style. Has several groups. On the phone, off, Available, Away etc. Managers can scroll the mouse over the user and see what call they are on etc. This is very helpfull because you dont have to program the phone. You have all the info right on the desktop. (as of now it is just a monitoring program. When we have time we will make it so the user can dial from outlook or by typing in the number on this program). Regards, Dovid --- Rob McKrill [EMAIL PROTECTED] wrote: Using the Buddy Watch functionality on the IP601 you can watch up to 6 people. The expansion modules are not good for much more than speed dials due to this limitation. After talking to our vendor, the reason it is limited to 6 is due to the current version of Asterisk Business Edition's lack of [documented/advertised] support for SUBSCRIBE/NOTIFY. Since Polycom is certified against ABE and the most recent release of ABE doesn't support this functionality, Polycom will not open their firmware up to allow more than six. I am hoping someone from Digium is monitoring this thread and that they might comment on when the new edition of ABE will be released so that we can actually utilize the full capabilities of the IP601's attendant consoles. Right now they (the attendant consoles) are pretty useless to me. Has anyone else had any success with them? Saul Diaz wrote: Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It works beautifull and not issues. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Web interface
Generaly you get what you pay for (with very few exceptions such as asterisk). Also as far as a web interface goes its really one that you get used to and like. There are lots out there. You goto find one that works for you. --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote: +++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. And unlike AMP, it is non-free. BTW: there is also DeStar: http://destar.berlios.de/ . Version 0.1.1 was recently released. Nice and clean. Generally runs its own daemon, though can run under apache. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preventing Asterisk from transfering the call
A)When you say stop asterisk from transfering the call what do you mean ? oNot to send it to VM if the user is away ? B)I think it depends on the phone. I know with the Polycoms you can program it directly in to the phone. (Done it in the past). --- Bartosz Piec [EMAIL PROTECTED] wrote: A user has set in his phone to transfer each call to another number. Is it possible to configure Asterisk not to transfer the calls? Or is it only phone setting? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
i've tested it with this config files and i worked:
extensions.conf
exten = 55,1,Dial(SIP/2271,20)
sip.conf
[2271]
type=friend
host=dynamic
secret=sip
allow=all
qualify=200
nat=no
Instead of 2271 you can put whatever you want.
good luck.
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with outbound call but there is
no
inbound call to my sip phone. the only message I get when I call from pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def [EMAIL PROTECTED] wrote:
there is no error message coming up on the pbx for in-bound calls (there
is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain
name
bindport=5060 ; UDP Port to bind to (SIP standard port
is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT SUPPORT
nat=no ; Global NAT settings (Affects all peers
and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls
Hi all,
I read many posts on asterisk mail site and been trying many different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601,
Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
phones to anywhere else but not receive phone calls. I can see the phones
on
sip show registry and sip show peers but no track phone calls for sip.
can you please shed some light on me how to go about solving this
problem?
thank you and best regards,
Ama
HR SIZE=1 Do you Yahoo!?
Re: [Asterisk-Users] RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3
Totally uneducated guess: If your version has the _expression_ parser, it has the leak. On 1/30/06, Damon Estep [EMAIL PROTECTED] wrote: Does anyone know what date this memory leak was introduced and/or how tocheck source code for it? I am running a pre-1.2 CVS head version and would like to know if thepotential problem exists. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
