Re: [asterisk-users] LDAPget or something else?
Hello David, * Klaverstyn, David C [EMAIL PROTECTED] [09-05-07 09:40]: We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there was something better. Are people using LDAPget or something else? I have ported LDAPget 2.0 to FreeBSD, works fine for me with asterisk 1.4. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spandsp-0.0.3 and asterisk 1.2
Hello Garth, * Garth van Sittert [EMAIL PROTECTED] [13-04-07 01:27]: Has anyone managed to get Asterisk 1.2 faxes working reliably with spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with a Digium b410p card. Everything compiled smoothly but only about 70% of faxes come through ok. Debugging shows nothing more than: app_rxfax.c: Fax receive not successful - result (11) Unexpected message received. The files are only 8 bytes long??? i have here spandsp-0.0.2.p26 asterisk-1.2.13_4 on FreeBSD with a HFC-S card and it works perfectly. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12
Hello Darryl, * Darryl Dunkin [EMAIL PROTECTED] [03-04-07 12:56]: November? It's DD/MM/ in his case, not MM/DD/. Either way, even two days is more than enough for me. is the format not? MM/DD/ DD.MM. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?
Hi, Zeeshan Zakaria schrieb: Grandstream phones are cheap because they use cheap stuff to manufacture them, plus their software/firmware and remote provisioning and configuration system is very basic. Their firmware was bad until newest version, and still some features like day light saving doesn't work. Will you be changing time on 80 phones twice a year? Let me repeat, their remote provisioning and configuration system is very basic, don't use for 80 phones installation, or you'll regret later. I use the phone only at home and it syncs fantastic with my ntp server, so changing time is not necessary. The phone is excellent for its price. And configuration and update via tftp works fine. The only negative point I have is that you must set an option in the config of the phone that is accepts the TFTP server from the dhcp server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.
Hi Matthew, Matthew Mackes (Webmail) schrieb: Zulty WIP 2- THESE PHONES ARE AWESOME!!! AWESOME!!! WiFi SIP phones- is it possible to provide a phonebook to this phones (via LDAP, TFTP, XML-file or anything else)? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing mpg123 with madplay under Solaris?
Hi Frank, Frank Tarczynski schrieb: The madplay executable works find on this box from the command line but is giving a segmentation fault when called from Asterisk. Has anyone already done this switch? Can they share some pointers? I run here Asterisk on FreeBSD with the buildin MOH from asterisk. It plays here mp3s perfectly, why not use it too? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Hello Steve, * Steve Underwood [EMAIL PROTECTED] [31-08-06 23:41]: Why would it not be a good idea to do things in software? hm, I have no idea Ok, I configured asterisk to receive the fax and wrote a small script which sent me the fax as pdf via email. Seems to work. Lets see how stable it is. The next days I will try to send mails with the help of a mail2fax gateway. Thx for all the answers. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with asterisk?
Hi, I use here mgetty+sendfax with a modem to receive and send fax messages. Is it possible to receive and send a fax with asterisk directly? I have two passive ISDN card (HFC-S chipset, one in NT mode the other in TE-mode) and a old ELSA Microlink modem via serial on my computer. The OS is FreeBSD. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Hello Roger, * Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]: did google for asterisk and fax show no results? yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is GXP2000 with latest firmware
Hello shadowym, * shadowym [EMAIL PROTECTED] [31-08-06 09:05]: I'm talking an office environment and not for in your kids room or anything like that. I use the phone only at home but this version works fine for me. I have no lockups, the phone runs now absolutely stable. No problem since the release date. Everything works fine from hints to receive calls, doing calls, MWI etc. The phone is configured via tftp. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP-2000 update to betafirmware?
Hi, currently I use version 1.1.0.16 for my GXP-2000 which works really fantastic. The only drawback I see is the addressbook. Is the firmware 1.1.1.9 stable enough to use the phone in normal environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000 says that there it is possible to download the addressbook as a XML-file. The problem is if the version not works it is not possible to downgrade to 1.1.0 Thx for any feedback, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP-2000 auf Betafirmware updaten?
Hi, currently I use version 1.1.0.16 for my GXP-2000 which works really fantastic. The only drawback I see is the addressbook. Is the firmware 1.1.1.9 stable enough to use the phone in normal environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000 says that there it is possible to download the addressbook as a XML-file. The problem is if the version not works it is not possible to downgrade to 1.1.0 Thx for any feedback, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] In CDR record not what I want
Hi,
I have the following rules:
exten = 4441,1,NoOp(--- ${CALLERID} calling on capi-extern (${EXTEN}) ---)
exten = 4441,2,Goto(dialin-privat,s,1)
exten = 4441,3,Hangup
[dialin-privat]
; Log incoming calls
exten = s,1,LDAPget(CALLERIDNAME=daheim)
exten = s,2,NoOP(--CALLERID=-${CALLERID}-, CALLERIDNUM=-${CALLERIDNUM}-,
EXTEN=-${EXTEN}--)
...
my CDR records says now that a call from unkown to s happened.
Is it possible that in the CDR record the number which has been called
is saved and not s?
e.g.
number unkown called 4441
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
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Re: [asterisk-users] In CDR record not what I want
Hello Rushowr,
* Rushowr [EMAIL PROTECTED] [11-08-06 06:10]:
It's because the standard CDR engine uses the last ${EXTEN} value as the
destination number
thx for that info I have rewritten now the section as a macro and now
everything works as expected.
Best regards,
Matthias
--
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build bigger and better idiot-proof programs, and the universe trying to
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Re: [asterisk-users] gxp-2000 configure line appearances
Hello Cavanna,, * Cavanna, Richard [EMAIL PROTECTED] [27-07-06 15:59]: The real thing that would help is a complete list of the configurable comands on the latest firmware so I can create the config file. try that config file, works perfectly for me. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ## Configuration template for GXP-2000 firmware version 1.0.2.13 ## ## Advanced/System-wide Options ## # Admin password for web interface P2 = admin # Silence Suppression. 0 - no, 1 - yes P50 = 1 # Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) P37 = 2 # Layer 3 QoS (IP Diff-Serv or Precedence value for RTP) P38 = 48 # Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP) P51 = 0 # Layer 2 QoS. 802.1p priority value (0 - 7) P87 = 0 # No Key Entry Timeout. Default - 4 seconds. P85 = 4 # Use # as Dial Key (if set to Yes, # will function as the (Re-)Dial key). 0 - no, 1 - yes P72 = 1 # Local RTP port (1024-65535, default 5004) P39 = 5004 # Use Random Port. 0 - no, 1 - yes P78 = 0 # Keep-alive interval (in seconds. default 20 seconds) P84 = 20 # Use NAT IP. This will enable our SIP client to use this IP in the SIP message. Example 64.3.153.50. P101 = # STUN server P76 = #- # Firmware Upgrade #- # Firmware Upgrade. 0 - TFTP Upgrade, 1 - HTTP Upgrade. P212 = 0 # Firmware Server Path P192 = 192.168.0.251 # Config Server Path P237 = 192.168.0.251 # Firmware File Prefix P232 = # Firmware File Postfix P233 = # Config File Prefix P234 = # Config File Postfix P235 = # Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is No. # When set to Yes(1), it will override the configured provision path and method. P145 = 0 # Automatic Upgrade. 0 - No, 1 - Yes (checking every defined days). Default is No. P194 = 1 # Check for new firmware every () minutes, unit is in minute, default is 7 days. P193 = 10080 # Use firmware pre/postfix to determine if f/w is required # 0 = Always Check for New Firmware # 1 = Check New Firmware only when F/W pre/suffix changes P238 = 0 # DTMF Payload Type P79 = 101 # Syslog Server (name of the server, max length is 64 charactors) P207 = 192.168.0.251 # Syslog Level (Default setting is NONE) # 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR P208 = 0 # NTP Server P30 = 192.168.0.251 # Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No. # When set to Yes(1), it will override the configured NTP server. P144 = 0 # Distinctive Ring Tone # Use custom ring tone 1 if incoming caller ID is the following: P105 = # Use custom ring tone 2 if incoming caller ID is the following: P106 = # Use custom ring tone 3 if incoming caller ID is the following: P107 = # Disable Call Waiting. 0 - no, 1 - yes P91 = 0 # Lock Keypad Update. 0 - no, 1 - yes P88 = 0 # Primary Account (Account 1) Settings # Account Active (In Use). 0 - no, 1 - yes P271 = 1 # Account Name P270 = # SIP Server P47 = sip.mycompany.com # Outbound Proxy P48 = proxy.mycompany.com # SIP User ID P35 = 8000 # Authenticate ID P36 = 8000 # Authenticate password P34 = # Display Name (John Doe) P3 = # Use DNS SRV. 0 - No, 1 - Yes. P103 = 0 # SIP User ID is phone number. 0 - no, 1 - yes P63 = 0 # SIP Registration. 0 - no, 1 - yes P31 = 1 # Unregister On Reboot. 0 - no, 1 - yes P81 = 0 # Register Expiration (in minutes. default 1 hour, max 45 days) P32 = 60 # Local SIP port (default 5060) P40 = 5060 # SIP T1 Timeout. RFC 3261 T1 value (RTT estimate) # 50 - 0.5 sec, 100 - 1 sec, 200 - 2 sec. Default 100. P209 = 100 # SIP T2 Interval. RFC 3261 T2 value. The maximum retransmit interval for non-INVITE requests and INVITE responses. # 200 - 2 sec, 400 - 4 sec, 800 - 8 sec. Default 400. P250 = 400 # NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive P52 = 0 # SUBSCRIBE for MWI. (Whether or not send SUBSCRIBE for Message Waiting Indication) 0 - No, 1 - Yes. P99 = 1 # Proxy-Require (A SIP extension to enable firewall penetration) P197 = # Voice Mail UserID (User ID/extension for 3rd party voice mail system) P33 = 88 # Send DTMF. 0 - in audio, 1 - via RTP, 2 - via SIP INFO P73 = 2 # Early Dial (use Yes
Re: [asterisk-users] Germany VOIP provider
Hello Thameem, * Thameem Ansari [EMAIL PROTECTED] [21-07-06 11:36]: I would like to get some details about voip providers in local germany. I am moving to germany and looking for some unlimited land+mobile minutes from provider. I also need a german DID with unlimited inbound and flat monthly rate. If anyone know anything, please reply. not everything you wanted but have a look at sipport.de. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Writing own applications for asterisk - read CALLERIDNUM
Hi,
I'm not sure if this is the right list for this question...
I have written a small application which looks up a number in a
database and return the name for the number if available.
But I have now the problem that I cannot read the variable CALLERIDNUM
from my script.
I tried it with:
value = pbx_builtin_getvar_helper(chan, key);
where chan is the value given from the initial function call (struct
ast_channel *chan) and key is set to CALLERIDNUM.
But the function always returns NULL.
The function is called with:
exten = 205,1,NoOP(--CALLERID=${CALLERID}, CALLERIDNUM=${CALLERIDNUM},
EXTEN=${EXTEN}--)
exten = 205,2,myAppGet(resolveToName)
exten = 205,3,Dial(SIP/201)
exten = 205,4,Hangup
The NoOp displays the right values.
TIA
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning. --
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Re: [asterisk-users] Writing own applications for asterisk - read CALLERIDNUM
Hello Russell, * Russell Bryant [EMAIL PROTECTED] [20-07-06 08:12]: ast_verbose(The channel cid num is: %s\n, chan-cid.cid_num); thx a lot! Everything is working perfectly now. :) Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] best hardphone for Asterisk?
Hi, Cullin J. Wible wrote: We have also deployed a dozen of the Linksys SPA-1001 single-line FXS adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy to deploy - $60-$70 US each. I bought a Grandstream GXP-2000 and played now a little with it. It seems to work really perfect. The Quality is compareable to my tiptel195 ISDN phone. The configuration can be done via TFTP and Web. I ordered it in Germany and the price was 97 EUR inclusive shipping. The only disadvantage the phone has is the very basic addressbook, but I think it will be improved with the next firmware versions. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing sound before dialing
Hi Tigran and Steve, Steve Totaro wrote: If you converted a wav file to gsm in the sounds directory, you have to delete the original wav file. Not that this is the issue in your case but just something I have run across. thx a lot, there was really a problem with the sound file. I have recorded it again with: Record(test2.gsm) and now it is working fine. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playing sound before dialing
Hi,
I have configured asterisk now with ENUM lookups which are working
really perfect.
Now I want to play a small soundfile before dial the number to inform
the caller which protocl is used (SIP, IAX2 or ISDN).
How can I do this?
With Playback it doesn't seems to work:
[iax2-sipport-out]
; with leading 3 using IAX-sipport
exten = s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out)
exten = s,2,Answer
exten = s,3,Playback(forwarded-iax)
exten = s,4,Dial(IAX2/portunity-out/${DIALSTR},,trRg)
exten = s,5,return
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning. --
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Re: [Asterisk-Users] GXP-2000 addressbook
Hi Gareth, Gareth Blades wrote: No I dont believe so. The address book is a new feature as it is very basic in my opinion and even editing it on the phone is difficult. I would expect a web based editing feature to be implemented at some point and once that is done it should be possible to do a mass update of the phones. ah ok, then I will wait for a new firmware :) Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 and Configdownload via TFTP
Hi, i got my Grandstream GXP-2000 phone today and want to configure it with TFTP. I downloaded the firmware 1.1.0.13 and put it into my tftp-server directory. Then I downloaded the template from: http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_Unix/Grandstream_Configuration_File_Template_1.0.6.x.txt renamed it to cfgmac-address Did the configuration in the new file and rebooted my phone. I can see in the log file from my tftp server that all files are loaded, the phone did a firmware upgrade. But it doesn't seems that the configuration file is loaded. Is it necessary to define on any place something that the phone use the config-file via tftp? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
Hi, I was now successful in getting syslog messages. Syslog says the following: Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099 GET cfgMAC What does errorcode 4099 mean? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
Hi Gareth, Gareth Blades wrote: You need to run the java based tool from the grandstream website to convert the template to a format the phone understands. thx that was the problem. Now it works fine. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 addressbook
Hi, is it possible to have one central phonebook and install it on the phone or using ldap? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Hi, is it possible to update the phonebook of the gxp-2000 via tftp? So I can maintain the phonebook central or using ldap etc.? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Hello Mimmus, * Mimmus [EMAIL PROTECTED] [07-06-06 16:52]: At first, I tried some chinese phones (AtCom) and they were a disaster. you talking ybout this phone? http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2 Has anyone some experience with this phone? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Hello Mimmus, * Mimmus [EMAIL PROTECTED] [07-06-06 17:20]: Yes good to known. I played with the idea to buy one of these. You would suggest GrandStream then? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 dialin with portunity
Hi, * Matthias Fechner [EMAIL PROTECTED] [03-06-06 22:33]: [portunity-in] type=user context=incoming-portunity permit=82.139.223.1/255.255.255.255 disallow=all allow=ulaw sry that doesn't help. ok correction, after forcing it to ulaw it is working fine. Thx a lot! Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 dialin with portunity
Hi, Matthias Fechner wrote: [portunity-in] type=user context=incoming-portunity permit=82.139.223.1/255.255.255.255 now I have the next problem. I can connect an iax phone and a sip phone to my asterisk. The problem is with incoming phone calls. If I use xlite everything is working perfectly but diax and idefisk are not working. So I think it is a problem with the IAX2 configuration. I got the call but i cannot hear anything and the calling person cannot her me. If I transfer the call to hold the calling person can hear MoH. Here is the debug log from asterisk: ___BEGIN___ Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 5ms SCall: 00058 DCall: 0 [82.139.223.1:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 03062006 CALLING NAME: diax0.9.15a LANGUAGE: en FORMAT : 2 CAPABILITY : 64798 ADSICPE : 0 DATE TIME : 2006-06-03 17:00:04 -- Accepting UNAUTHENTICATED call from 82.139.223.1: requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 1ms SCall: 1 DCall: 00058 [82.139.223.1:4569] FORMAT : 4 -- Executing Dial(IAX2/portunity-out-1, IAX2/idefixSIP/idefix) in new stack -- Called idefix Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 4ms SCall: 6 DCall: 0 [192.168.0.151:4569] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw|alaw|gsm) CALLING NUMBER : 03062006 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: diax0.9.15a LANGUAGE: de USERNAME: idefix FORMAT : 14 CAPABILITY : 63502 ADSICPE : 0 DATE TIME : 2006-06-03 17:00:06 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 4ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 00058 DCall: 1 [82.139.223.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 00016ms SCall: 00310 DCall: 6 [192.168.0.151:4569] FORMAT : 14 -- Call accepted by 192.168.0.151 (format unknown) -- Format for call is (gsm|ulaw|alaw) Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 6 DCall: 00310 [192.168.0.151:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 3ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 6 DCall: 00310 [192.168.0.151:4569] -- IAX2/idefix-6 is ringing Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 4ms SCall: 1 DCall: 00058 [82.139.223.1:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 4ms SCall: 00058 DCall: 1 [82.139.223.1:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02000ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: PONG Timestamp: 02000ms SCall: 6 DCall: 00310 [192.168.0.151:4569] RR_JITTER : 0 RR_LOSS : 0 RR_PKTS : 1 RR_DELAY: 40 RR_DROPPED : 0 RR_OUTOFORDER : 0 Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 02000ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 03485ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 03485ms SCall: 6 DCall: 00310 [192.168.0.151:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 03488ms SCall: 00310 DCall: 6 [192.168.0.151:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 03488ms SCall: 6 DCall: 00310 [192.168.0.151:4569] -- IAX2/idefix-6 answered IAX2/portunity-out-1 -- Attempting native bridge of IAX2/portunity-out-1 and IAX2/idefix-6 -- Operating with different codecs 4[(ulaw)] 14[(gsm|ulaw|alaw)] , can't native bridge... Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: CONTROL Subclass: (255?) Timestamp: 03489ms SCall: 1 DCall: 00058 [82.139.223.1:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 001 Type: CONTROL Subclass: ANSWER Timestamp: 03492ms SCall: 1 DCall: 00058
[Asterisk-Users] Asterisk 1.2.8
Hi, is a new port for Asterisk 1.2.8 for FreeBSD out? Regarding to the changelog there some bugs fixed with iax and the codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved. Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.8
Hi, * Matthias Fechner [EMAIL PROTECTED] [03-06-06 22:13]: is a new port for Asterisk 1.2.8 for FreeBSD out? Regarding to the changelog there some bugs fixed with iax and the codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved. sry, mail should go to [EMAIL PROTECTED] Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 dialin with portunity
Hello Tim, * Tim Panton [EMAIL PROTECTED] [03-06-06 19:12]: You have a weird codec problem. Try changing the iax config to limit it to ulaw and see if that helps: [portunity-in] type=user context=incoming-portunity permit=82.139.223.1/255.255.255.255 disallow=all allow=ulaw sry that doesn't help. You might also want to upgrade to asterisk 1.2.8 - which has some fixes in the IAX code - but I don't know if any are related to this - I haven't had a chance to install it yet. ah great if FreeBSD port is up-to-date I will upgrade and give some feedback here. Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 and dialin
Hi, after some corrections in my settings IAX2 dialin seems to work now. I get the incoming call, but i cannot here anything or can speak. (If I take the call the other side see that the connection is established if I close the call the other site is seeing it too) If I press hold in Idefisk the other side can hear MoH but not me. Asterisk print in the CLI interface that he starts MoH. The firewall isn't blocking any incoming or outgoing package. (I cannot find anything in the log and every blocked package will be logged) My setup is, FreeBSD6 going online with ppp, NAT is done with pf and the firewall too. Asterisk is configured to bind to 0.0.0.0, so it should bind to my tun0 interface and the external IP. netstat -an says: udp4 0 0 *.4569 *.* udp4 0 0 *.5060 *.* If I call outside everything is working fine. Is this a problem with NAT or the maybe the firewall or is it necessary to change some configoptions in asterisk? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 dialin with portunity
Hello Joshua, Joshua Colp wrote: [portunity-out] type=friend host=iax.iaxport.de username=XXX secret=YY context=incoming-portunity notransfer=yes Only if the username is specified as portunity-out when the other side dials you. Otherwise your Asterisk has no idea what to authenticate them as so it takes a guess and in the end settles on guest. But should not asterisk here see, that the call is comming in from the host: host=iax.iaxport.de or from the username=iaxXX? In the SIP configuration I do it this way. Or need I to define some other parameters in the section [portunity-out] or easily rename it. If I get a call, asterisk says the following: (I hope everything is in :) ) ___CUT___ Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 5 DCall: 0 [82.139.223.1:4569] VERSION : 2 CALLED NUMBER : matthiasfechner CODEC_PREFS : (ulaw|alaw|gsm) CALLING NUMBER : [EMAIL PROTECTED] CALLING PRESNTN : 1 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Matthias Fechner LANGUAGE: de USERNAME: iaxX FORMAT : 14 CAPABILITY : 63502 ADSICPE : 0 DATE TIME : 2006-05-30 11:50:48 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 4ms SCall: 00084 DCall: 5 [82.139.223.1:4569] AUTHMETHODS : 3 CHALLENGE : 182242807 USERNAME: iaxX Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00032ms SCall: 5 DCall: 00084 [82.139.223.1:4569] MD5 RESULT : 872efd005c628f31f74c2b142ca05cb5 Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00028ms SCall: 22848 DCall: 4 [192.168.0.151:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00022ms SCall: 00084 DCall: 5 [82.139.223.1:4569] FORMAT : 14 -- Call accepted by 82.139.223.1 (format unknown) -- Format for call is (gsm|ulaw|alaw) Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00022ms SCall: 5 DCall: 00084 [82.139.223.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 00049 DCall: 0 [82.139.223.1:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : [EMAIL PROTECTED] CALLING NAME: Matthias Fechner LANGUAGE: de FORMAT : 2 CAPABILITY : 64798 ADSICPE : 0 DATE TIME : 2006-05-30 11:50:50 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 6 DCall: 00049 [82.139.223.1:4569] -- Accepting UNAUTHENTICATED call from 82.139.223.1: requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 8ms SCall: 6 DCall: 00049 [82.139.223.1:4569] FORMAT : 4 ___CUT___ Best regards, Matthias Fechner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 dialin with portunity
Hello Joshua, * Joshua Colp [EMAIL PROTECTED] [30-05-06 12:41]: happen as you might expect. You need to use permit and deny lines to get the user entry matched. Check into the sample iax.conf to see how to do this. thx a lot! The following entry in iax.conf is doing the trick: [portunity-in] type=user context=incoming-portunity permit=82.139.223.1/255.255.255.255 Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Define call-groups
Hi,
i want to define some call groups like:
extensions.conf
[globals]
GROUP1=IAX2/idefixSIP/200
[capi-in]
exten = 55,1,Dial(${GROUP1})
exten = 55,2,Hangup
But Dial will not dial the defined numbers.
exten = 55,1,Dial(IAX2/idefixSIP/200) works fine.
What is here wrong?
Or is there a better way available to do it?
Best regards,
Matthias
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[Asterisk-Users] Problem with IAX2 dialin with portunity
Hi,
I'm using http://www.portunity.net/
I configured now asterisk with the following setup:
iax.conf:
register = XXX:[EMAIL PROTECTED]
[portunity-out]
type=friend
host=iax.iaxport.de
username=XXX
secret=YY
context=incoming-portunity
notransfer=yes
[guest]
type=user
context=default
;callerid=Guest IAX User
And in extensions.conf:
[default]
;exten = s,1,DIAL(IAX2/idefix)
exten = s,1,NoOp(--- ${CALLERID} calling on portunity over IAX2
(${EXTEN}) ---)
exten = s,2,Set(LANGUAGE()=de)
exten = s,3,Ringing
exten = s,4,Wait,4
exten = s,5,Answer
exten = s,6,Playback(invalid)
exten = s,7,Hangup
[incoming-portunity]
;exten = s,1,DIAL(IAX2/idefix)
exten = s,1,NoOp(--- ${CALLERID} calling on portunity over IAX2
(${EXTEN}) ---)
exten = s,2,Set(LANGUAGE()=de)
exten = s,3,Ringing
exten = s,4,Wait,4
exten = s,5,Answer
exten = s,6,Playback(invalid)
exten = s,7,Hangup
But if I get a call it is always matched with the section [guest] and
calls the context default. But if a call is incoming from portunity
should not the section [portunity-out] with context incoming-portunity
be called?
iax2 show registry says:
Host UsernamePerceived Refresh State
82.139.223.1:4569 XX.XX.12.XX:4569 60 Registered
Thx a lot for help!
Best regards,
Matthias
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Re: [Asterisk-Users] Calling a person over Internet
Hello Michiel, * Michiel van Baak [EMAIL PROTECTED] [27-05-06 17:15]: You have to do a couple of things: 1. Open your firewall so it allows the protocol you want to use. ok, that should be easy. 2. Configure asterisk to accept guest calls 3. Configure asterisk to ring some phones when someone dials your domain. and how is this working? Is the person who want call me dial [EMAIL PROTECTED] How can ppl reach me if they only use SIP? If there is a site or howto etc. available it would be a pleasure for my to get something to read :) Remember, only ppl with voip can reach you this way. Normal landline phones can only reach you when you have a landline connected to a tdm card or if you connect with a voip provider. I have a ISDN card in my PC which is working perfectly. Thx for answers. Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling a person over Internet
Hi, i have now asterisk running at home. Is it possible that a other person can call me now with my domain or must I use a VoIP provider? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
