[asterisk-users] Need help in communicating H323 and SIP
Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=1 tcpEnd=2 ;UDP udpStart=1 udpEnd=2 [general] port = 1720 bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. These flags ; are used in the generation of call detail records. ; ;amaflags = default ; ; You may specify a default account for Call Detail Records in addition ; to specifying on a per-user basis ; ;accountcode=lss0101 ; ; You can fine tune codecs here using allow and disallow clauses ; with specific codecs. Use all to represent all formats. ; ;disallow=all ;allow=all ; turns on all installed codecs ;disallow=g723.1; Hm... Proprietary, don't use it... ;allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw ; see doc/rtp-packetization for framing options ; ; User-Input Mode (DTMF) ; ; valid entries are: rfc2833, inband ; default is rfc2833 ;dtmfmode=rfc2833 ; ; Default RTP Payload to send RFC2833 DTMF on. This is used to ; interoperate with broken gateways which cannot successfully ; negotiate a RFC2833 payload type in the TerminalCapabilitySet. ; ; You may also specify on either a per-peer or per-user basis below. ;dtmfcodec=101 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; IP address or Host name - The acutal IP address or hostname of your GK gatekeeper = DISABLE ;gatekeeper=10.142.17.68 ; ; ; Tell As terisk whether or not to accept Gatekeeper ; routed calls or not. Normally this should always ; be set to yes, unless you want to have finer control ; over which users are allowed access to Asterisk. ; Default: YES ; ;AllowGKRouted = yes ; ; When the channel works without gatekeeper, there is possible to ; reject calls from anonymous (not listed in users) callers. ; Default is to allow anonymous calls. ; ;AcceptAnonymous = yes ; ; Optionally you can determine a user by Source IP versus its H.323 alias. ; Default behavour is to determine user by H.323 alias. ; ;UserByAlias=no ; ; Default context gets used in siutations where you are using ; the GK routed model or no type=user was found. This gives you ; the ability to either play an invalid message or to simply not ; use user authentication at all. ; ;context=default ; ; Use this option to help Cisco (or other) gateways to setup backward voice ; path to pass inband tones to calling user (see, for example, ; http://www.cisco.com/warp/public/788/voip/ringback.html https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html ) ; ; Add PROGRESS information element to SETUP message sent on outbound calls ; to notify about required backward voice path. Valid values are: ; 0 - don't add PROGRESS information element (default); ; 1 - call is not end-end ISDN, further call progress information can ;possibly be available in-band; ; 3 - origination address is non-ISDN (Cisco accepts this value only); ; 8 - in-band information or an appropriate pattern is now available; ;progress_setup = 3 ; ; Add PROGRESS information element (IE) to ALERT message sent on incoming ; calls to notify about required backwared voice path. Valid values are: ; 0 - don't add PROGRESS IE ( default); ; 8 - in-band information or an appropriate pattern is now available; ;progress_alert = 8 ; ; Generate PROGRESS message when H.323 audio path has established to create ; backward audio path at other end of a call. ;progress_audio = yes ; ; Specify how to inject non-standard information into H.323 messages. When ; the channel receives messages with tunneled information, it automatically ; enables the same option for all further outgoing messages independedly on ; options has been set by the configuration. This behavior is required, for ; example, for Cisco CallManager when Q.SIG tunneling is enabled for a ; gateway where Asterisk lives. ; The option can be used multiple times, one option per line. ;tunneling=none ; Totally disable tunneling (default) ;tunneling=cisco; ; Enable Cisco-specific tunneling ;tunneling=qsig ; Enable tunneling via Q.SIG messages ; ;-- JITTER BUFFER
Re: [asterisk-users] wireless VOIP phone recommendations?
Chris Bagnall schrieb: - No shared adress book (especially it should be shared between phone on different base stations). I can access an online adress book, but only the built in, and you cannot set up your own online book. You can send address books to the phone in standard vcard format (though for some reason it insists on getting them in PC line break format rather than unix format). For client deployments with a significant number of Gigasets we tell them to update the phonebook on a web interface, then hit publish which pushes it out to each handset using a simple curl call. Yeah, i am aware of this, but if you have a great number of phones and many base stations you have to access the web interface quite often. We worked on a scripted way of erasing the phone books and uploading the new data, but the incorporated the risk of breaking the script if an firmware update changes the web interface. The builtin phone book if rather limited. 170 entrys is easy reached for a company phone book. But maybe someday there will be a really useful solution, like accessing a ldap server. - Does not listen to SIP Message Call completed elsewhere. If you let several phones ring for an incoming call, and it get answered at one phone, all the others will have a missed call in there list, this isn't quite true. Over the day this list fills up and you don't know if there really is a missed call among them. Does asterisk provide this SIP message? Looking around at the collection of Snom 370s and 320s here, all of them claim to have varying number of missed calls from when the call's been answered from another in the ring group. Or is there perhaps a config setting to enable this I've not spotted? Well, actually it does ... After patching it in ... I have found the patch in the Bug Tracker and it works, if the telephone listens to it. Regards. -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
- Original Message From: Olivier [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 February, 2008 7:35:10 AM Subject: Re: [asterisk-users] External MWI question for Asterisk Do you send those notifications to SIP hardphones ? Then, how do you proceed ? Is there a standard way to make (or stop) a SIP hardphone Message Waiting Indicator blinking ? Hi Olivier, Yes we send MWI notifications to all user agents on our service which include IP Phones and ATAs. We use the externnotify option in voicemail.conf to pipe out to an external script everytime a voicemail is checked. That script does a few things that allows our MWI service to know that there are no longer any new voicemails for a specific and a NOTIFY request is sent to the user agent to turn the MWI off. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
Femi wrote: Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls That largely depends on whether you need to do transcoding and between which codecs, etc. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cosini iAN7s
Hi, Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk? Please let me know how it performed and what issues you faced Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Scalability
Hi, Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transferring a call received by an agent in a queue
Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten = 11000,1,Dial(SIP/11000,,t) exten = 1001,1,Dial(SIP/1001,,t) exten = 1002,1,Dial(SIP/1002,,t) exten = 1003,1,Dial(SIP/1003,,t) exten = 1004,1,Dial(SIP/1004,,t) exten = 2001,1,agi,login.php exten = 2002,1,Queue(FAO|tT) exten = 2004,1,MusicOnHold exten = 2004,2,Hangup When I call from 11000 to 1001, I can press # and type 2004 to transfer and 11000 gets MOH. When I dial 2002 (queue) from 11000, 1001 rings and I am able to talk both ways, but nothing happens when I press # at 1001. No logs appears at asterisk console in verbose 3 level. I am using asterisk 1.4.15. All the docs indicate that I just need to invoke Queue application with tT to enable call transfer. But that does not seems to work in my case. queues.conf [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [FAO] musiconhold = default strategy = roundrobin servicelevel = 60 eventmemberstatus = yes eventwhencalled = yes timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm sip.conf [general] context=from-sip allowguest=no bindport=5060 bindaddr=192.168.3.36 srvlookup=yes [11000] host=dynamic type=friend dtmfmode=RFC2833 username=11000 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no [1001] host=dynamic type=friend dtmfmode=RFC2833 username=1001 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no Thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preventing IAX frame concatenation
On 8 Feb 2008, at 00:39, David Hogan wrote: Alternatively you could fix the client :-) Heh :) Although it's a situation that won't happen in (our) production, for the sake of completeness I'll probably upgrade the client. Actually, it does (assuming you guys still run Tesco's UK service). The first frame in a GSM call is often 66 bytes, but there is only one of it and it is at the beginning of the call, so it's hardly a major problem. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
This will be closed service provider network with own VoIP phones and gateways so we can assume that there is no transcoding Femi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 08 February 2008 12:15 To: Asterisk Users Subject: Re: [asterisk-users] Asterisk Scalability Femi wrote: Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls That largely depends on whether you need to do transcoding and between which codecs, etc. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cosini iAN7s
Cosini SS7 work for 99% of all cases. Femi ha scritto: Hi, Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk? Please let me know how it performed and what issues you faced Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
That's very unfortunate. I use now a workaround : I'm just switching (with gotos) between extensions and use some macros but always within the same context. I'll try to remeber it for next time :) Cheers, Yves. On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote: On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: * Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). Yes, unfortuneately that's the thing you have to do. You have to add each context you want - in static conf file like this: [db_na] switch = Realtime/db_na [db_busy] switch = Realtime/db_busy You can have as many extensions you like with whatever commands, but contexts still should be registered. Generally editing and debugging of complete dialplan in DB is not so easy - so you should keep your main logic in static, but use realtime for data that actually changes. Regards, Atis * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite option - gona have problems?
Hi list, can anyone tell me how problematic it is setting canreinvite=yes ? I know if its to avoid issues with bad implementatins of SIP on other devices then maybe you cant give a black and white answer, but any constructive comments welcome! Reason being I think I have to set this to yes to enable mediaproxy RTP proxy on my OpenSER box to interoperate correctly with Asterisk, Thanks Andy.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: That's very unfortunate. I use now a workaround : I'm just switching (with gotos) between extensions and use some macros but always within the same context. Well, you should create contexts for your main features, and you can write few of them in extensions.conf - it's a small trouble when compared to gain from separation of different functionality. Regards, Atis I'll try to remeber it for next time :) Cheers, Yves. On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote: On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: * Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). Yes, unfortuneately that's the thing you have to do. You have to add each context you want - in static conf file like this: [db_na] switch = Realtime/db_na [db_busy] switch = Realtime/db_busy You can have as many extensions you like with whatever commands, but contexts still should be registered. Generally editing and debugging of complete dialplan in DB is not so easy - so you should keep your main logic in static, but use realtime for data that actually changes. Regards, Atis * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk versions (newbie)
Vytenis, As far as I understand it 1.2 zaptel should be used with 1.2 Asterisk, 1.4 with 1.4. As for 1.2 vs 1.4, it depends on if you want new features and any bug-fixes. 1.2 is a closed project (I think). Just compile from source if its not available as an RPM in 1.4 for Debian. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vytenis Sabaliauskas Sent: 08 February 2008 15:13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Question about Asterisk versions (newbie) Hello, I would like to consulate with you guys. I'm setting up an Asterisk server on Debian. The problem is that Rhino drivers are only compatible with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel 1.2, and that suits our needs. Is there a bleeding need to use latest version of asterisk? I have managed to install Asterisk 1.4 and Zaptel 1.2 but then i got the ioctl error for chan_zap. Is combination Asterisk 1.4 and Zaptel 1.2 possible? In forums I've red that its somehow bad :) Thanks for any info -- V. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GS/* phonebook
Hello, I wrote a small AJAX phonebook targeted at Grandstream phones although the basic functionality doesn't require a SIP-phone. Asterisk integration (call history, incoming call info, click-to-dial) is not yet implemented, but on my ToDo list. Features * Licenced under the GPL v2 * Uses gettext. So translation should be easy. Includes German and English. * Uses xajax for a nice interactive feel * Multi-user capable. Phonebook entries are assigned a group and only members of this group can see the entry. * Can serve auto-generated XML phonebooks to Grandstream phones. * User-selectable ring tones. No more messing around in the TFTP- or HTTP-server files. Each user can select which three ring tones her/his phone should download. If somebody is interested: http://almosthappy.de/gsphonebook/ Lars -- Home is where .emacs is. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
We are building a Ruby on Rails interface that we will probably put out as open source later this spring. I worked around this problem with #exec statements. This is what is in my extensions.conf file: #exec /path/to/scripts/load_extensions.rb This runs a Ruby script that rips through my extensions table and builds a context with the appropriate switch statement for each realtime context. The output is included much like the text from an include file. Of course, it still requires a reload if you add a brand new context. regards, Mike Clark Yves Räber wrote: That's very unfortunate. I use now a workaround : I'm just switching (with gotos) between extensions and use some macros but always within the same context. I'll try to remeber it for next time :) Cheers, Yves. On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote: On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: * Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). Yes, unfortuneately that's the thing you have to do. You have to add each context you want - in static conf file like this: [db_na] switch = Realtime/db_na [db_busy] switch = Realtime/db_busy You can have as many extensions you like with whatever commands, but contexts still should be registered. Generally editing and debugging of complete dialplan in DB is not so easy - so you should keep your main logic in static, but use realtime for data that actually changes. Regards, Atis * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interoperability between TE412P and Eurotech PRI E1 GSM CDMA Gateway
Hi, I am about to purchase an Eurotech PRI E1 GSM CDMA Gateway to operate with my Asterisk's TE412P interface. Anyone here has any experience of having this combination? Any success or failure stories would be greatly appreciated. Thanks in advance. Ash ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Permission denied when obtaining Status
Greetings, I've set up the AMI and am able to authenticate, however I am unable to execute Action: Status. I get a permission denied error: asterisk:/etc/asterisk# telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action: Login Username: myusername Secret: mypassword Response: Success Message: Authentication accepted Action: Status Response: Error Message: Permission denied I then get expected Event / displayconnects output. The user information from manager.conf is as follows: [myusername] secret = mypassword deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 read = system,call,log,verbose,command,agent,user write = none displayconnects = yes Any idea why I am getting permission denied when executing Action: Status? Cheers, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Permission denied when obtaining Status
Steve Shepherd wrote: Greetings, I've set up the AMI and am able to authenticate, however I am unable to execute Action: Status. I get a permission denied error: *snipped read = system,call,log,verbose,command,agent,user write = none without the ability to 'write' a command, you are in fact 'denied' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
Don't forget to 1000,1,Answer the call Moj John Von Essen wrote: Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk versions (newbie)
On Fri, Feb 08, 2008 at 05:12:33PM +0200, Vytenis Sabaliauskas wrote: Hello, I would like to consulate with you guys. I'm setting up an Asterisk server on Debian. The problem is that Rhino drivers are only compatible with Zaptel 1.2. Thats seems odd to me. Are you sure? I would suggest you to check again. By default debian stable offers asterisk 1.2 and zaptel 1.2, and that suits our needs. Is there a bleeding need to use latest version of asterisk? I have managed to install Asterisk 1.4 and Zaptel 1.2 but then i got the ioctl error for chan_zap. Is combination Asterisk 1.4 and Zaptel 1.2 possible? In forums I've red that its somehow bad :) One thing to realize is that there are also quite some changes in the 1.2 series . Specifically, around zaptel 1.2.13 . Another thing to realize is that asterisk build chan_zap according to zaptel.h found at /usr/include/zaptel/zaptel.h (where it was installed by zaptel 1.4 , zaptel 1.2 installs it at /sur/include/linux/zaptel.h) . So my guess is that you have headers from zaptel 1.4 installed on the system. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk using C
Soumya Kat wrote: Hi, I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8 system. Asterisk works fine for me and I can log into Asterisk-GUI and monitor asterisk. What I would like to know is how to get information such as SIP users, number of SIP connections and traffic associated with those from asterisk using a C Code. Thank you. Soumya, This may help: http://www.voip-info.org/wiki-Asterisk+manager+API Not sure what you mean by traffic though. For call history, you might look at: http://www.voip-info.org/wiki/view/CDR For current status of sip lines, etc. the Asterisk Manager Interface (AMI) is still your friend. AMI command SipPeers will force all event packets to be issued for sip peers which you can catch and analyze. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk using C
Soumya Kat wrote: Hi, I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8 system. Asterisk works fine for me and I can log into Asterisk-GUI and monitor asterisk. What I would like to know is how to get information such as SIP users, number of SIP connections and traffic associated with those from asterisk using a C Code. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Although the code is pretty messy, you can see how I got this sort of information from asterisk for my monitor project AstSee. Source is available at http://www.astsee.com/ Mojo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk versions (newbie)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino Drivers are agnostic to the version of zaptel you are using with 1small exception. You can build any of our drivers against zaptel up to 1.4.7 without any patching or fancy foot work. You can guild our 2.2.3beta2 and when released the 2.2.3 drivers against all zaptel versions including 1.4.8 our drivers expect the kernel headers or source a c compiler and zaptel to be at /usr/src/zaptel zaptel should be built and installed to ensure all the zaptel.h etc are in their expected homes. - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHrKCodloC7YyaIOoRAtblAJ9alZhyHHGef/6tnK14sg5sejsGWQCeN3fU yXZ2Mwr2lvbTVDO3SEhSIM4= =9ELw -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor Asterisk using C
Hi, I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8 system. Asterisk works fine for me and I can log into Asterisk-GUI and monitor asterisk. What I would like to know is how to get information such as SIP users, number of SIP connections and traffic associated with those from asterisk using a C Code. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 Echo
After Andrew's suggestion, if that isn't the problem, spend some more time on OSLEC to be darn sure it's operating properly -- that thing works like a champ for my crappy lines! Moj Brent Davidson wrote: We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the new system and they are both reporting echo on both incoming and outgoing calls. The echo seems to be confined to the Snom 300 phones and is not heard by the person on the zap line. The echo is only the voice of the person using the Snom phone. There doesn't seem to be any echo of the analog line audio. I have tried adjusting the gain of the lines, turning on echo cancellation, Turning on echo training and nothing seems to work. At one of the branches, I re-compiled asterisk and zaptel using the OSLEC drivers and that doesn't seem to have had any effect either. What am I missing? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
Thanks! Femi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: 08 February 2008 15:20 To: asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Scalability Hi! Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls Use the Wiki, Luke! http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning Cheers, Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk versions (newbie)
Hello, I would like to consulate with you guys. I'm setting up an Asterisk server on Debian. The problem is that Rhino drivers are only compatible with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel 1.2, and that suits our needs. Is there a bleeding need to use latest version of asterisk? I have managed to install Asterisk 1.4 and Zaptel 1.2 but then i got the ioctl error for chan_zap. Is combination Asterisk 1.4 and Zaptel 1.2 possible? In forums I've red that its somehow bad :) Thanks for any info -- V. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dealipedia
Lol - so I read about this website today - www.dealipedia.com http://www.dealipedia.com/ And I thought cool, lets start typing in a few names of companies I know who have taken funding recently. Check out the username of the person who submitted the Fonality deal - http://www.dealipedia.com/deal_view_investment.php?r=2838 I guess this is a comment about their thoughts on open source software being used for commercial purposes. And yes seeing I know you are all going to go and type in Digium - here it is as well http://www.dealipedia.com/deal_view_investment.php?r=101 Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Domainname for outgoing uri-dialing
Hi,
I use outgoing URI-dialing for my sip-phones as suggested in
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
The relevant extensions look like this:
[dial-uri]
exten = _[a-z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,1,Macro(uridial,[EMAIL PROTECTED])
[macro-uridial]
exten = s,1,Set(dialuri=${CUT(ARG1,\;,1)})
exten = s,n,Set(CALLERID(number)=${CALLERID(number)[EMAIL PROTECTED])
exten = s,n,Dial(SIP/${dialuri},120,tr)
exten = s,n,Congestion()
I end up with an outgoing SIP-Invite with contact and from-headers like
[EMAIL PROTECTED]@IP-address
That obviously is not what I want. I can set the fromdomain value in the
general-part of my sip.conf and leave away the setting of the callerid
which fixes the problem. But as I want to use different domains for the
outgoing calls depending on the user, that is not a solution for me. Can
I influence the generation of the outgoing domainname somehow?
thanks for your help
Bjoern
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[asterisk-users] Rejected calls to Sylantro server
I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and outbound calls. Has anyone made this work in the past? This is the rejection sent by the switch at the other end: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.1.4.110:5060;received=67.55.x.x;branch=z9hG4bK71d476d2;rport=5060 From: 515XXX sip:[EMAIL PROTECTED];tag=as6a5cd6c8 To: sip:[EMAIL PROTECTED];tag=aprqngfrt-n2bk9j1c6 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asking for recommendations on Asterisk Boxes or Appliances
I believe trixbox can fulfill your requierements. regards - Mensaje original De: Paul Hales [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviado: jueves 7 de febrero de 2008, 22:45:00 Asunto: Re: [asterisk-users] Asking for recommendations on Asterisk Boxes or Appliances Astlinux on your own built box? PaulH On Thu, 2008-02-07 at 14:11 -0800, John Constalgie wrote: Hi there, I am looking to buy an Asterisk Appliance or Box for my organization and was hoping to ask for recommendations. My ideal box is a small device in size like Digium's AA50 Asterisk Appliance ( http://www.digium.com/en/products/appliance/ ) but will still have these technical features : - Run Linux obviously - Run a fullly configurable distro of Asterisk (not embedded) - Can support a decent number of concurrent calls ( = 50 ) - Support AGI (preferably Perl or PHP) - Preferably has Text-To-Speech and IVR Does anyone have any recommendations for Asterisk Appliances like that? I believe the AA50 cannot run AGI scripts and uses a skimmed down version of Asterisk Thanks John __ Helping your favorite cause is as easy as instant messaging. You IM, we give. Learn more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tarjeta de crédito Yahoo! de Banco Supervielle. Solicitá tu nueva Tarjeta de crédito. De tu PC directo a tu casa. www.tuprimeratarjeta.com.ar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wait in Queue for 120 seconds for agent A to become free, THEN ring next agent
Don't worry - I paste this leink becaus eyou should have e good understanding about what the queue() cmd does to be safe in implementation phase: http://www.voip-info.org/wiki-Asterisk+cmd+Queue See also: http://astrecipes.net/index.php?n=118 Thanks l. On Tue, 05 Feb 2008 06:31:16 +0100, Kev S [EMAIL PROTECTED] wrote: Sorry to be painful, But how do i set the queue timeout? Regards Kev -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: * Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). Yes, unfortuneately that's the thing you have to do. You have to add each context you want - in static conf file like this: [db_na] switch = Realtime/db_na [db_busy] switch = Realtime/db_busy You can have as many extensions you like with whatever commands, but contexts still should be registered. Generally editing and debugging of complete dialplan in DB is not so easy - so you should keep your main logic in static, but use realtime for data that actually changes. Regards, Atis * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
This is standard stuff. I have switch over 200 simultaneous with g711 on a 1-U, Xeon-DualCore @ 3.0 using RH versions of Linux. Even higher with pass thru (no-transcoding) on g729. ..mike.. At 07:54 AM 2/8/2008, Femi wrote: This will be closed service provider network with own VoIP phones and gateways so we can assume that there is no transcoding Femi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 08 February 2008 12:15 To: Asterisk Users Subject: Re: [asterisk-users] Asterisk Scalability Femi wrote: Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls That largely depends on whether you need to do transcoding and between which codecs, etc. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wireless VOIP phone recommendations?
2008/2/8, Tobias Wolf [EMAIL PROTECTED]: Chris Bagnall schrieb: - No shared adress book (especially it should be shared between phone on different base stations). I can access an online adress book, but only the built in, and you cannot set up your own online book. You can send address books to the phone in standard vcard format (though for some reason it insists on getting them in PC line break format rather than unix format). For client deployments with a significant number of Gigasets we tell them to update the phonebook on a web interface, then hit publish which pushes it out to each handset using a simple curl call. Yeah, i am aware of this, but if you have a great number of phones and many base stations you have to access the web interface quite often. We worked on a scripted way of erasing the phone books and uploading the new data, but the incorporated the risk of breaking the script if an firmware update changes the web interface. The builtin phone book if rather limited. 170 entrys is easy reached for a company phone book. But maybe someday there will be a really useful solution, like accessing a ldap server. - Does not listen to SIP Message Call completed elsewhere. If you let several phones ring for an incoming call, and it get answered at one phone, all the others will have a missed call in there list, this isn't quite true. Over the day this list fills up and you don't know if there really is a missed call among them. Does asterisk provide this SIP message? Looking around at the collection of Snom 370s and 320s here, all of them claim to have varying number of missed calls from when the call's been answered from another in the ring group. Or is there perhaps a config setting to enable this I've not spotted? Well, actually it does ... After patching it in ... I have found the patch in the Bug Tracker and it works, if the telephone listens to it. I couldn't find this patch in Bug Tracker. Do you have a number or a description of this SIP message (is it anINVITE option ?) Cheers Regards. -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
Hi! Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls Use the Wiki, Luke! http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning Cheers, Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade 1.2 - 1.4 voice files
Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need good voicemail documentation
Jaap Winius wrote: * Why can't I delete any voicemail messages? (Response: Message undeleted.) * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response: I'm sorry, I did not understand your response.) * How come if I put [EMAIL PROTECTED] in my phone's context of sip.conf, do I get an error? (CLI: ...Remote host can't match request NOTIFY to call...) I don't think you will find any of these in an asterisk voicemail documentation project. You need to examine the CLI with sufficient verbosity, and ask for our help if you don't understand what's in there. These are all problems that would be VERY atypical to have with asterisk's voicemail. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending a message from inside voicemailmain.
As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I do seem to remember that in 1.2, it wasn't possible to send a message to ones-self. This bug fix apparently corrects that situation. Well, I guess it would, if only it were possible to send a message to anyone. When I try to send the message, the system silently fails, returning me to the prompt that would be spoken after the message was successfully sent. On the console, i get: [Feb 8 22:12:10] WARNING[19921]: app_voicemail.c:2850 leave_voicemail: No entry in voicemail config file for '3825' If I try to send a message to a mailbox that actually doesn't exist, Allison tels me about it, as one would expect. I don't get the error message on the console. I am told what file is being played, again, as expected. Any ideas? -- Bill in Denver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [asterisk-biz]SIP to SIP professional community
Hello Everyone, I am currently operating a VoIP business in Canada and joined the list. I have seen some very useful ideas and information posted daily in this forum. I have also noticed that there are user who barter, sell, trade services, products, etc. Wonderful !!. What I do notice as well is, none have created a directory for affordable voice communications for users of the list to really call each other. My business platform is geared for all forms of communications in the Asterisk world and I would like to create a worldwide SIP to SIP community for the user list. How it work is very simple: 1. You would be given a member number where members can call you world wide for free !!! 2. Post your member # with every posting so you can be reached off list. 3. You can also have a off list conference call as well. If you are keen on this idea please let me know. It would be nice to hear voices instead of writing back and forth. All members worldwide welcome. Yours truly, Nigel Dennis 281 617 1465 http://www.cellcallback.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending a message from inside voicemailmain.
On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW +1-303-722-7209 wrote: As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. So does this work if you use the directory, if you don't use the directory, or neither? Is the mailbox you're sending to within the same context? Are you using the 'default' context or one of your own? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] oneway audio with asterisk behind cisco pix 506
Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the fixups disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) [incoming] exten = 4025901000,1,Goto(1000,1) exten = 1000,1,Goto(customer_ivr,s,1) Thanks sunMoonstar. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the “fixups” disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) [incoming] exten = 4025901000,1,Goto(1000,1) exten = 1000,1,Goto(customer_ivr,s,1) Thanks sunMoonstar. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the fixups disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) [incoming] exten = 4025901000,1,Goto(1000,1) exten = 1000,1,Goto(customer_ivr,s,1) Thanks sunMoonstar. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending a message from inside voicemailmain.
On Fri, 8 Feb 2008, Tilghman Lesher wrote: On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW +1-303-722-7209 wrote: As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. So does this work if you use the directory, if you don't use the directory, or neither? Is the mailbox you're sending to within the same context? Are you using the 'default' context or one of your own? Hi, Thanks for mentioning contexts. All of us are in the default context. So I started playing around with the options pertaining to contexts. I found that if I uncommented searchcontexts=yes, I could send from inside. The explanation says that if the parameter is set to no, only the default context will be searched, which should have worked for me. By setting it to yes, I now have lots of happy users. Thanks again. -- Bill in Denver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Ravi, I will explain changing the config in asterisk and the pix: Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 1 to 10050 (to start, you will need to increase later as ports fill up) (use insert to make a change in a file) to save: 1. esc 2. shift + colon 3. wq (to save) If you made a mistake and do not want to save but you changed something in the file: 1. esc 2. shift + colon 3. q! (to exit) Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the static and conduit commands so this is a example from my setup. Theses are not usable IPs on the Internet or my IPs but just an example outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254) dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254) interface ethernet0 100full (sets the duplex and turns on interface) interface ethernet1 100full (sets the duplex and turns on interface) nameif ethernet0 outside security0 ( lower security) nameif ethernet1 dmz security50 (higher security) no fixup protocol sip 5060 no fixup protocol sip udp 5060 ! - this makes things easier so now the pix knows the IP of the asterisk box and maps the ip to the name just for configuration purposes only so if you had 20 servers or devices you wanted public access to it's just easier to remember their names versus IPs. name 192.168.254.11 dns name 192.168.254.10 asterisk ! - the static command is used as a permanent mapper from one inside, dmz, or other to the global ip vice versa. (Rule of thumb if you map using static make sure you have a conduit command) static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0 ! - here is where you open the ports on the global side to the asterisk box. (the conduit command allows connections from lower security interfaces to higher security interfaces) conduit permit udp host 192.168.1.22 eq 1 any conduit permit udp host 192.168.1.22 eq 10001 any conduit permit udp host 192.168.1.22 eq 10002 any conduit permit udp host 192.168.1.22 eq 10003 any conduit permit udp host 192.168.1.22 eq 10004 any conduit permit udp host 192.168.1.22 eq 10005 any Hope this helps! --Otis Ravichandran Rajagopal wrote: Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the fixups disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) [incoming] exten = 4025901000,1,Goto(1000,1) exten = 1000,1,Goto(customer_ivr,s,1) Thanks sunMoonstar. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
LOL I guess all I was asking for the changes to be made in the Cisco PIX 506. I think you gave me a short tutorial on VI as well. Thanks once again for this help. Let me work on these changes and test the one-way audio problem and go from there. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:55 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, I will explain changing the config in asterisk and the pix: Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 1 to 10050 (to start, you will need to increase later as ports fill up) (use insert to make a change in a file) to save: 1. esc 2. shift + colon 3. wq (to save) If you made a mistake and do not want to save but you changed something in the file: 1. esc 2. shift + colon 3. q! (to exit) Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the static and conduit commands so this is a example from my setup. Theses are not usable IPs on the Internet or my IPs but just an example outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254) dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254) interface ethernet0 100full (sets the duplex and turns on interface) interface ethernet1 100full (sets the duplex and turns on interface) nameif ethernet0 outside security0 ( lower security) nameif ethernet1 dmz security50 (higher security) no fixup protocol sip 5060 no fixup protocol sip udp 5060 ! - this makes things easier so now the pix knows the IP of the asterisk box and maps the ip to the name just for configuration purposes only so if you had 20 servers or devices you wanted public access to it's just easier to remember their names versus IPs. name 192.168.254.11 dns name 192.168.254.10 asterisk ! - the static command is used as a permanent mapper from one inside, dmz, or other to the global ip vice versa. (Rule of thumb if you map using static make sure you have a conduit command) static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0 ! - here is where you open the ports on the global side to the asterisk box. (the conduit command allows connections from lower security interfaces to higher security interfaces) conduit permit udp host 192.168.1.22 eq 1 any conduit permit udp host 192.168.1.22 eq 10001 any conduit permit udp host 192.168.1.22 eq 10002 any conduit permit udp host 192.168.1.22 eq 10003 any conduit permit udp host 192.168.1.22 eq 10004 any conduit permit udp host 192.168.1.22 eq 10005 any Hope this helps! --Otis Ravichandran Rajagopal wrote: Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the fixups disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten =
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
No problem. :-P I thought it might wise to include everything you needed just in case!! LOL! You are welcome!!! --Otis Ravichandran Rajagopal wrote: LOL I guess all I was asking for the changes to be made in the Cisco PIX 506. I think you gave me a short tutorial on VI as well. Thanks once again for this help. Let me work on these changes and test the one-way audio problem and go from there. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:55 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, I will explain changing the config in asterisk and the pix: Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 1 to 10050 (to start, you will need to increase later as ports fill up) (use insert to make a change in a file) to save: 1. esc 2. shift + colon 3. wq (to save) If you made a mistake and do not want to save but you changed something in the file: 1. esc 2. shift + colon 3. q! (to exit) Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the static and conduit commands so this is a example from my setup. Theses are not usable IPs on the Internet or my IPs but just an example outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254) dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254) interface ethernet0 100full (sets the duplex and turns on interface) interface ethernet1 100full (sets the duplex and turns on interface) nameif ethernet0 outside security0 ( lower security) nameif ethernet1 dmz security50 (higher security) no fixup protocol sip 5060 no fixup protocol sip udp 5060 ! - this makes things easier so now the pix knows the IP of the asterisk box and maps the ip to the name just for configuration purposes only so if you had 20 servers or devices you wanted public access to it's just easier to remember their names versus IPs. name 192.168.254.11 dns name 192.168.254.10 asterisk ! - the static command is used as a permanent mapper from one inside, dmz, or other to the global ip vice versa. (Rule of thumb if you map using static make sure you have a conduit command) static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0 ! - here is where you open the ports on the global side to the asterisk box. (the conduit command allows connections from lower security interfaces to higher security interfaces) conduit permit udp host 192.168.1.22 eq 1 any conduit permit udp host 192.168.1.22 eq 10001 any conduit permit udp host 192.168.1.22 eq 10002 any conduit permit udp host 192.168.1.22 eq 10003 any conduit permit udp host 192.168.1.22 eq 10004 any conduit permit udp host 192.168.1.22 eq 10005 any Hope this helps! --Otis Ravichandran Rajagopal wrote: Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the fixups disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include =
Re: [asterisk-users] Sending a message from inside voicemailmain.
On Friday 08 February 2008 23:28:01 William F. Acker WB2FLW +1-303-722-7209 wrote: On Fri, 8 Feb 2008, Tilghman Lesher wrote: On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW +1-303-722-7209 wrote: As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. So does this work if you use the directory, if you don't use the directory, or neither? Is the mailbox you're sending to within the same context? Are you using the 'default' context or one of your own? Thanks for mentioning contexts. All of us are in the default context. So I started playing around with the options pertaining to contexts. I found that if I uncommented searchcontexts=yes, I could send from inside. The explanation says that if the parameter is set to no, only the default context will be searched, which should have worked for me. By setting it to yes, I now have lots of happy users. Well, I spent a couple hours and tracked this down. Basically what was happening was that we were passing a literal context of (null), which is why the mailbox wasn't being found (that's the string that you get when you printf a NULL). And due to this string, it took a long time to figure out why. This is now fixed in revision 103197 for SVN 1.4. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Otis, I wanted to clarify what you said and what I comprehended. the SIP protocols are disabled in fixup. Having said that I guess all I have to do is just the following. the inside IP of asterisk server is 192.168.5.0 On the cisco PIX firewall enter the following. 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any conduit permit udp host ... . 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any conduit permit udp host in the rtp.conf in /etc/asterisk change the ending port 2 (which is what it currently is) to 10050 Is there an easier way to make the entries in Cisco PIX firewall ? Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Saturday, February 09, 2008 12:18 AM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 No problem. :-P I thought it might wise to include everything you needed just in case!! LOL! You are welcome!!! --Otis Ravichandran Rajagopal wrote: LOL I guess all I was asking for the changes to be made in the Cisco PIX 506. I think you gave me a short tutorial on VI as well. Thanks once again for this help. Let me work on these changes and test the one-way audio problem and go from there. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:55 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, I will explain changing the config in asterisk and the pix: Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 1 to 10050 (to start, you will need to increase later as ports fill up) (use insert to make a change in a file) to save: 1. esc 2. shift + colon 3. wq (to save) If you made a mistake and do not want to save but you changed something in the file: 1. esc 2. shift + colon 3. q! (to exit) Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the static and conduit commands so this is a example from my setup. Theses are not usable IPs on the Internet or my IPs but just an example outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254) dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254) interface ethernet0 100full (sets the duplex and turns on interface) interface ethernet1 100full (sets the duplex and turns on interface) nameif ethernet0 outside security0 ( lower security) nameif ethernet1 dmz security50 (higher security) no fixup protocol sip 5060 no fixup protocol sip udp 5060 ! - this makes things easier so now the pix knows the IP of the asterisk box and maps the ip to the name just for configuration purposes only so if you had 20 servers or devices you wanted public access to it's just easier to remember their names versus IPs. name 192.168.254.11 dns name 192.168.254.10 asterisk ! - the static command is used as a permanent mapper from one inside, dmz, or other to the global ip vice versa. (Rule of thumb if you map using static make sure you have a conduit command) static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0 ! - here is where you open the ports on the global side to the asterisk box. (the conduit command allows connections from lower security interfaces to higher security interfaces) conduit permit udp host 192.168.1.22 eq 1 any conduit permit udp host 192.168.1.22 eq 10001 any conduit permit udp host 192.168.1.22 eq 10002 any conduit permit udp host 192.168.1.22 eq 10003 any conduit permit udp host 192.168.1.22 eq 10004 any conduit permit udp host 192.168.1.22 eq 10005 any Hope this helps! --Otis Ravichandran Rajagopal wrote: Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (1 to 10200) unless you write a script to just copy and paste about 1 to 2 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and
Re: [asterisk-users] External MWI question for Asterisk
2008/2/8, Grey Man [EMAIL PROTECTED]: - Original Message From: Olivier [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 February, 2008 7:35:10 AM Subject: Re: [asterisk-users] External MWI question for Asterisk Do you send those notifications to SIP hardphones ? Then, how do you proceed ? Is there a standard way to make (or stop) a SIP hardphone Message Waiting Indicator blinking ? Hi Olivier, Yes we send MWI notifications to all user agents on our service which include IP Phones and ATAs. We use the externnotify option in voicemail.conf to pipe out to an external script everytime a voicemail is checked. That script does a few things that allows our MWI service to know that there are no longer any new voicemails for a specific and a NOTIFY request is sent to the user agent Do you mean your script does send a NOTIFY messages to hardphones ? Then, how did you write such SIP-aware script (language, ...) ? If not, how external script and Asterisk do communicate ? to turn the MWI off. Regards, Regards Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing SIP server user extension... Dial string issue...
Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:[EMAIL PROTECTED]); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102 Extensions.conf entry [context-gs102] exten = s,1, Answer(); exten = s,n, Playback(demo-congrats); exten = s,n, Meetme(8600051); exten = 1234,1, Answer(); exten = 1234,n, Playback(demo-congrats); exten = 1234,n, Meetme(8600051); When I dial I get following error on console -- Executing Dial(SIP/331-6263, SIP/gs102:[EMAIL PROTECTED]) in new stack -- Called gs102:[EMAIL PROTECTED] -- SIP/192.168.2.81-0343 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/331-6263, ) in new stack == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/331-6263' I want to call extension 1234 defined under gs102 defined context-gs102 context... what should be the exact Dialed SIP URL ? -ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Note also that if you point to the DNS name rather than the IP address of the asterisk server on the phones trying to register, you can set NAT=NO on the asterisk side and the sip FIXUP command on the PIX will handle everything correctly making this workaround unnecessary - Original Message - From: Ravichandran Rajagopal [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 8, 2008 8:54:23 PM (GMT-0800) America/Los_Angeles Subject: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the “fixups” disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include = customer exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [agnosco] include = local-extensions include = customer_ivr include = incoming [customer_ivr] include = local-extensions exten = s,1,Answer exten = s,n,Background(agnosco_intro) exten = s,n,WaitExten ;Dial said extensions exten = 5,1,Dial(SIP/[EMAIL PROTECTED],30) [incoming] exten = 4025901000,1,Goto(1000,1) exten = 1000,1,Goto(customer_ivr,s,1) Thanks sunMoonstar.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
