Re: [asterisk-users] Voicemail issue
https://issues.asterisk.org/jira/browse/18998 https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 _ From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure unused BRI ports from a HA8 board ? [SOLVED]
2011/6/15 A J Stiles asterisk_l...@earthshod.co.uk On Wednesday 15 Jun 2011, Olivier wrote: At the moment, my console is full with messages such as : [Jun 15 09:06:25] WARNING[2140]: chan_dahdi.c:3369 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! [Jun 15 09:06:25] WARNING[2141]: chan_dahdi.c:3369 pri_find_dchan: No D-channels available! Using Primary channel 12 as D-channel anyway! These messages relate to ports 3 and 4 which are unused and unplugged. I can connect those 2 ports to each other (using a straight RJ11-RJ11 cable ?) and tie them to an used DAHDI group but this wouldn't work for a single port and I'm wondering if a smarter solution exists. Suggestions ? Make sure the entries for these ports are commented out in /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf . Great : this did it ! Usually, I'm using dahdi_genconf to generate Dahdi config but I must say I couldn't manage to generate NT configs or to disable some ports with it (could you ?). Thanks for helping. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - dialog-info+xml - NAT
Hi, I try to solve my problem with asterisk and BLF function. I have registered peers from realtime with subscriptions but only type is mwi (shown by 'sip show subscriptions'). Peers are registered from behind the NAT - may it be the cuase why they not subscribed with dialog-info+xml? Regards, Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial out conference
Hi, You may used the Page() function of asterisk. Which will work the same as you are required at this moment. On Wed, Jun 15, 2011 at 12:51 PM, Alex Balashov abalas...@evaristesys.comwrote: On 06/15/2011 01:34 AM, Nikhil wrote: Hi Asterisk support dialout conference?.My requirement is when type a CLI command with argument as a number ,asterisk should able to make a call to that number and when connected ,that channel should entered in to the conference room,like this I should able to add multiple users into the conference.I am using ConfBridge application for asterisk version 1.6.2 This is something that can be accomplished with the manager interface or call files. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
On Tue, 14 Jun 2011, Florent THOMAS wrote: Le 12/06/2011 20:41, Florent THOMAS a écrit : Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) I'll try it ASAP and let you know. Hy Gordon, I checked my devices and observed that I'm the owner of 2 LinkSys SPA2102-R configured by keyyo. I used the Voice interaction menu to reset the device. Unfortunately it stills asking me a password. I don't really know much about the SPA devices, sorry. (the only ATAs I've really used have been Grandstreams) It sounds like they might be locked into the keyyo service (if that's an ITSP who provided them pre-configured) Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
Le 15/06/2011 10:53, Gordon Henderson a écrit : On Tue, 14 Jun 2011, Florent THOMAS wrote: Le 12/06/2011 20:41, Florent THOMAS a écrit : Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) I'll try it ASAP and let you know. Hy Gordon, I checked my devices and observed that I'm the owner of 2 LinkSys SPA2102-R configured by keyyo. I used the Voice interaction menu to reset the device. Unfortunately it stills asking me a password. I don't really know much about the SPA devices, sorry. (the only ATAs I've really used have been Grandstreams) It sounds like they might be locked into the keyyo service (if that's an ITSP who provided them pre-configured) Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You're totally right and actually I 'd just engaged a procedure to have the information from my IPST. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial out conference
Thanks for the helps I use channel originate command to achieve this. Command: asteriskCLI channel originate SIP/201 application ConfBrigde 1234 This will make a call to the 201 user and when connected,it will be routed to conference room . Thanks NIkhil On 06/15/2011 02:17 PM, virendra bhati wrote: Hi, You may used the Page() function of asterisk. Which will work the same as you are required at this moment. On Wed, Jun 15, 2011 at 12:51 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: On 06/15/2011 01:34 AM, Nikhil wrote: Hi Asterisk support dialout conference?.My requirement is when type a CLI command with argument as a number ,asterisk should able to make a call to that number and when connected ,that channel should entered in to the conference room,like this I should able to add multiple users into the conference.I am using ConfBridge application for asterisk version 1.6.2 This is something that can be accomplished with the manager interface or call files. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOICEMAIL CONFIGURATION
i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOICEMAIL CONFIGURATION
On 15 Jun 2011, at 11:20, mahesh katta wrote: i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG A lot of filesystems are case sensitive. Maybe you wrote your configuration in caps? This would also explain why you couldn't provide anything from the logs ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi 2.4.0 and Squeeze [SOLVED]
Is there a dahdi_cfg in your boot sequence? When I modify dahdi config files I always launch dahdi_cfg otherwise I get errors like yours. Giorgio On 06/14/2011 05:37 PM, Olivier wrote: After a reboot, I can't reproduce the problem anymore which is quite frustating. 2011/6/14 Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote: Hi, I'm using a two-years old installation script for the first time on a Squeeze (linux 2.6.32) platform. For an unknown reason (might be an obvious one), Dahdi can't be loaded anymore. 1. First of all, it seems /dev/dahdi content was previously (ie in Lenny) owned by asterisk:asterisk (asterisk is run as asterisk). Now it is owned by root. Any clue about this ? asterisk:asterisk ? How did you set the ownership? with a simple chown -R asterisk:asterisk /dev/dahdi/* Through a file in /etc/udev/rules/ ? 2. Secondly, I changed /dev/dahdi content ownership by hand. If the device files are not static (which they normally aren't) the device files will get re-generated next time the driver loads. Yes, it's true. Then when I'm trying to load chan_dahdi, I can read : module load chan_dahdi Unable to load module chan_dahdi Command 'module load chan_dahdi' failed. [Jun 14 15:41:53] WARNING[8150]: chan_dahdi.c:1469 dahdi_open: Unable to specify channel 1: No such device or address [Jun 14 15:41:53] ERROR[8150]: chan_dahdi.c:8816 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Jun 14 15:41:53] ERROR[8150]: chan_dahdi.c:14229 build_channels: Unable to register channel '1-2' Suggestions ? What is the output of lsdahdi? http://docs.tzafrir.org.il/dahdi-linux/#_procfs_interface_proc_dahdi -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com mailto:jabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir http://iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for helping (for both Andrew and Tzafrir). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call file challenge...
Hi, I think you need to update *waittime* parameter in .call file please put atleast 10 seconds. for more understanding please try to read *http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out* Regards Dhaval On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic positivelyoptimis...@gmail.com wrote: Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a Call failed to go through, reason (3) Remote end Ringing message when attempting to originate a call from a call file. Numbers changed to protect the innocent using call file //CALL FILE// Channel: DAHDI/g1/918005551212 Callerid: 8002211212 WaitTime: 2 MaxRetries: 6 RetryTime: 8 Context: xs-globx-ds3 Extension: 12564286000 Priority: 1 //CALL FILE// //CLI SNIPPET// -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 2) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:24] NOTICE[27177]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 3) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:34] NOTICE[27179]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 4) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:44] NOTICE[27182]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 5) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:54] NOTICE[27183]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 6) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:36:04] NOTICE[27185]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 7) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:36:14] NOTICE[27188]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing //CLI SNIPPET// Software Version(s) Asterisk 1.6.2.16.1 DAHDI Version: 2.4.0 libpri version: 1.4.11.5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOICEMAIL CONFIGURATION
Sir,
thanks for reply .
exten = 8501,1,VoicemailMain(s${CALLERIDNUM})
exten = 8501,2,Hangup
exten = 4578909,1,AGI(agi://127.0.0.1:4577/call_log)
exten =
4578909,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = 4578909,3,Dial(SIP/173,30,tTo)
exten = 4578909,4,Dial(Zap/g0/0559421629,,tTo)
exten = 4578909,5,VoiceMail(u173)
exten = 4578909,6,playback(vm-goodbye)
exten = 4578909,7,Hangup
exten = 4578909,104,VoiceMail(b173)
exten = 4578909,105,playback(vm-goodbye)
exten = 4578909,106,Hangup()
voicemail.conf
1001 = 1001,1001 Mailbox,
1002 = 1002,1002 Mailbox,
1003 = 1003,1003 Mailbox,
1004 = 1004,1004 Mailbox,
1005 = 1005,1005 Mailbox,
1006 = 1006,1006 Mailbox,
101 = 101,101 Mailbox,
111 = 111,111 Mailbox,
169 = 169,169 Mailbox,
170 = 170,170 Mailbox,
171 = 171,171 Mailbox,
172 = 172,172 Mailbox,
173 = 173,173 Mailbox,
174 = 174,174 Mailbox
this is my configuration . i am using vicidial.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Wed, Jun 15, 2011 at 4:01 PM, Steven Howes steve-li...@geekinter.netwrote:
On 15 Jun 2011, at 11:20, mahesh katta wrote:
i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT
VOICEMAIL. WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG
A lot of filesystems are case sensitive. Maybe you wrote your configuration
in caps? This would also explain why you couldn't provide anything from the
logs ;)
S
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Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!
Dears; The problem was related to something else. The Digium card has two PRI ports, actually to get it UP, I have to configure the two ports and both of those two ports to take the timing from span 1. Why this, I do not know ! Although I am using only one E1 connected to span 1, so why I have to configure the other span !! After configuring the second span, so now one D channel for span 1 is UP and the other is down (because no E1 cable connected to the other span), now I can remove the configuration for the other span and the D channel for the first span will stay UP, but at anytime, the E1 might come back down again and I have to configure the other span port again to get the E1 up on the first span. Any advise for this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CONFERENCE CONFIGURATION REQUIRE
Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact configuration in dialplan and any to edit confugration files please help me . and how can they cut the conference of after concall. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIGIUM PRI CARDS REQUIRE
Hi, I Required digium PRI cards, single span, dual span, quad core . so any body give me cotaion for this cards and I required also grandstream fxs/fxo devices . give me for this quotation . price and details.. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf and BRI NT spans in system.conf
On Tue, Jun 14, 2011 at 11:36:33PM +0200, Olivier wrote: Hi, My genconf_parameter is : # grep -v ^# genconf_parameters lc_countryfr context_linesremote group_lines1 bri_sig_stylebri echo_canoslec pri_termtype SPAN/*NT (I also tried various identations and syntax for pri_termtype line such as bri_termtype but I still get the same output file). But still, generated system.conf contains only TE spans. That parameter is intended for spans that dahdi_genconf cannot tell if they are TE or NT. It seems dahdi_genconf is sure that this specific port is TE. What device is it? # grep term,te system.conf span=1,1,0,ccs,ami,term,te span=2,2,0,ccs,ami,term,te span=3,3,0,ccs,ami,term,te span=4,4,0,ccs,ami,term,te Is that actually a generated line? Did i miss something ? If that matters, I'm using Dahdi 2.4.0 and Asterisk 1.6.1.18. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio after a reinvite changing codec
HI list, no idea?? :) M. On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.comwrote: Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UACASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g729 --- g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line: Oooh, we need to change our audio formats since our peer supports only g729 and asterisk send back 200 OK to the provider. At this point we have one way rtp audio: UACASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g711 --- g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!
What company card you have? Copy paste your dahdi config and chan_dahdi.conf -- Sent from my iPhone On Jun 15, 2011, at 6:53 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; The problem was related to something else. The Digium card has two PRI ports, actually to get it UP, I have to configure the two ports and both of those two ports to take the timing from span 1. Why this, I do not know ! Although I am using only one E1 connected to span 1, so why I have to configure the other span !! After configuring the second span, so now one D channel for span 1 is UP and the other is down (because no E1 cable connected to the other span), now I can remove the configuration for the other span and the D channel for the first span will stay UP, but at anytime, the E1 might come back down again and I have to configure the other span port again to get the E1 up on the first span. Any advise for this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf and BRI NT spans in system.conf
2011/6/15 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Jun 14, 2011 at 11:36:33PM +0200, Olivier wrote: Hi, My genconf_parameter is : # grep -v ^# genconf_parameters lc_countryfr context_linesremote group_lines1 bri_sig_stylebri echo_canoslec pri_termtype SPAN/*NT (I also tried various identations and syntax for pri_termtype line such as bri_termtype but I still get the same output file). But still, generated system.conf contains only TE spans. That parameter is intended for spans that dahdi_genconf cannot tell if they are TE or NT. It seems dahdi_genconf is sure that this specific port is TE. What device is it? It's a Digium HA8+B400M board. # lsdahdi ### Span 1: WCBRI/0/0 HA8- Board 1 (MASTER) AMI/CCS 1 BRIClear (In use) (SWEC: OSLEC) 2 BRIClear (In use) (SWEC: OSLEC) 3 BRIHardware-assisted HDLC (In use) ### Span 2: WCBRI/0/1 HA8- Board 1 AMI/CCS 4 BRIClear (In use) (SWEC: OSLEC) 5 BRIClear (In use) (SWEC: OSLEC) 6 BRIHardware-assisted HDLC (In use) ### Span 3: WCBRI/0/2 HA8- Board 1 AMI/CCS RED 7 BRIClear (SWEC: OSLEC) RED 8 BRIClear (SWEC: OSLEC) RED 9 BRIHardware-assisted HDLC RED ### Span 4: WCBRI/0/3 HA8- Board 1 AMI/CCS RED 10 BRIClear (SWEC: OSLEC) RED 11 BRIClear (SWEC: OSLEC) RED 12 BRIHardware-assisted HDLC RED I also tried with Digium B410P and Junghanns PCIe QuadBRI I can't remember ever seeing an NT configuration file, as if the pri_termtype statement was skipped. # grep term,te system.conf span=1,1,0,ccs,ami,term,te span=2,2,0,ccs,ami,term,te span=3,3,0,ccs,ami,term,te span=4,4,0,ccs,ami,term,te Is that actually a generated line? Yes, it's an extract from a generated file. Did i miss something ? If that matters, I'm using Dahdi 2.4.0 and Asterisk 1.6.1.18. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
This was a bug in 1.4, 1.6.x, and 1.8. It is fixed in the latest release of each of the Asterisk versions. Check the Changelog for 1.8.4, you might see the bugtracker ID with the patch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Tuesday, June 14, 2011 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
The same issue was present in 1.6 a few weeks ago and is fixed in latest 1.6. Maybe latest 1.8.4 does not have this issue. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Wednesday, June 15, 2011 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; brya...@zktech.com Subject: Re: [asterisk-users] Voicemail issue I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. _ From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 _ From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] change destination on digit
Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
The latest 1.8.x solved the problem for us on multiple servers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 15, 2011 9:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; brya...@zktech.com Subject: Re: [asterisk-users] Voicemail issue The same issue was present in 1.6 a few weeks ago and is fixed in latest 1.6. Maybe latest 1.8.4 does not have this issue. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Wednesday, June 15, 2011 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; brya...@zktech.com Subject: Re: [asterisk-users] Voicemail issue I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
Hi, it seems to be fixed in 1.8.4. At least I can't reproduce it there. Karsten Am Mittwoch, den 15.06.2011, 09:29 -0400 schrieb Mike: The same issue was present in 1.6 a few weeks ago and is fixed in latest 1.6. Maybe latest 1.8.4 does not have this issue. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Wednesday, June 15, 2011 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; brya...@zktech.com Subject: Re: [asterisk-users] Voicemail issue I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. __ From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 __ From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] connecting to SIP Provider with virtual IP from pacemaker cluster
Hello ! i am new to this list and asterisk. I run asterisk 1.4 on a OpenSuSE 11.4. My SIP Provider needs my IP to connect the local area number to my IP and also for there firewall. I plan to run asterisk in a pacemaker cluster that is not the problem and works. My problem is the virtual IP from the cluster and the connection to the SIP Provider. My Server has 2 NIC's one intern to register the VOIP Phone's and one external to register to the SIP Provider. If i run asterisk with a virtuail IP configuration i get problems to connect to the SIP Provider because the ip route uses the real Hardware NIC ip. OK this i can solve with setting a sorce IP with ip route. That works with ssh an other but not with asterisk. tcpdump always show the the ip from the real Hardware device. A bindaddr in the SIP conf is first not possible because i had to connect from internal and external. But when i try a bindaddr on the virtual device tcpdump shows the same result that the connection comes from the real IP. maybe some had the same problem an can give me a hint. BR/Torsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
Le 15/06/2011 11:06, Florent THOMAS a écrit : Le 15/06/2011 10:53, Gordon Henderson a écrit : On Tue, 14 Jun 2011, Florent THOMAS wrote: Le 12/06/2011 20:41, Florent THOMAS a écrit : Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) I'll try it ASAP and let you know. Hy Gordon, I checked my devices and observed that I'm the owner of 2 LinkSys SPA2102-R configured by keyyo. I used the Voice interaction menu to reset the device. Unfortunately it stills asking me a password. I don't really know much about the SPA devices, sorry. (the only ATAs I've really used have been Grandstreams) It sounds like they might be locked into the keyyo service (if that's an ITSP who provided them pre-configured) Gordon -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You're totally right and actually I 'd just engaged a procedure to have the information from my IPST. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hy gordon, My former IPST informed me that this kind of device can't be configured for another IPST. Have you already experinced this kind of behaviour? Do you know some devices that aren't so locked? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re connecting to SIP Provider with virtual IP, from pacemaker cluster
Hi, If your cluster's virtual IP is using ip aliasing (eg eth0:0), i think your problem come from UDP flows, they are, in opposition to TCP flows, unconnected, so the IP stack take the shortest route/interface to send them, wich is when this is the default route, the real interface and not the aliased. For exemple if you have eth0 the real, eth0:0 the virtual, you can try to add in your failback/failover cluster script something like this : # when the virtual ip come up ip r a "SIP_PROVIDER_IP" via "GATEWAY_IP" dev eth0:0 # when the virtual ip come down, maybe facultative because the route is deleted when the interface fall down ip r d "SIP_PROVIDER_IP" via "GATEWAY_IP" dev eth0:0 Regards, Cédric Le 15/06/11 19:01, asterisk-users-requ...@lists.digium.com a écrit : Date: Wed, 15 Jun 2011 17:28:36 +0200 From: rosenber...@taoweb.at Subject: [asterisk-users] connecting to SIP Provider with virtual IP from pacemaker cluster To: asterisk-users@lists.digium.com Message-ID: d265002689c5927864dc1da6ae1f1...@taoweb.at Content-Type: text/plain; charset=UTF-8; format=flowed Hello ! i am new to this list and asterisk. I run asterisk 1.4 on a OpenSuSE 11.4. My SIP Provider needs my IP to connect the local area number to my IP and also for there firewall. I plan to run asterisk in a pacemaker cluster that is not the problem and works. My problem is the virtual IP from the cluster and the connection to the SIP Provider. My Server has 2 NIC's one intern to register the VOIP Phone's and one external to register to the SIP Provider. If i run asterisk with a virtuail IP configuration i get problems to connect to the SIP Provider because the ip route uses the real Hardware NIC ip. OK this i can solve with setting a sorce IP with ip route. That works with ssh an other but not with asterisk. tcpdump always show the the ip from the real Hardware device. A bindaddr in the SIP conf is first not possible because i had to connect from internal and external. But when i try a bindaddr on the virtual device tcpdump shows the same result that the connection comes from the real IP. maybe some had the same problem an can give me a hint. BR/Torsten -- Cédric Lemarchand 52 avenue de l'Europe 78160 Marly-le-Roi France Tel. +33 (0)1 30 08 88 88 Cell. +33 (0)6 37 23 40 93 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!
The card is Digium card T2XXP (PCI) as I mentioned in my email. I added the configuration for the second port (span 2) to work, otherwise it does not work. I just added the below lines in the files system.conf and chan_dahdi.conf, all other lines are the default lines. The asterisk version is: Asterisk 1.8.3.2, DAHDI Version: 2.4.1 and libpri-1.4.11.5. system.conf: span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,crc4,yellow bchan=32-46,48-62 dchan=47 chan_dahdi.conf: context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=IncomingPSTN group=1 signalling=pri_cpe switchtype=euroisdn channel=32-46,48-62 So any advise why I have to configure the second span? I only need span 1. Regards Bilal What company card you have? Copy paste your dahdi config and chan_dahdi.conf -- Sent from my iPhone On Jun 15, 2011, at 6:53 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; The problem was related to something else. The Digium card has two PRI ports, actually to get it UP, I have to configure the two ports and both of those two ports to take the timing from span 1. Why this, I do not know ! Although I am using only one E1 connected to span 1, so why I have to configure the other span !! After configuring the second span, so now one D channel for span 1 is UP and the other is down (because no E1 cable connected to the other span), now I can remove the configuration for the other span and the D channel for the first span will stay UP, but at anytime, the E1 might come back down again and I have to configure the other span port again to get the E1 up on the first span. Any advise for this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
On Wed, 15 Jun 2011, Florent THOMAS wrote: Le 15/06/2011 11:06, Florent THOMAS a écrit : Le 15/06/2011 10:53, Gordon Henderson a écrit : On Tue, 14 Jun 2011, Florent THOMAS wrote: Le 12/06/2011 20:41, Florent THOMAS a écrit : Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) I'll try it ASAP and let you know. Hy Gordon, I checked my devices and observed that I'm the owner of 2 LinkSys SPA2102-R configured by keyyo. I used the Voice interaction menu to reset the device. Unfortunately it stills asking me a password. I don't really know much about the SPA devices, sorry. (the only ATAs I've really used have been Grandstreams) It sounds like they might be locked into the keyyo service (if that's an ITSP who provided them pre-configured) You're totally right and actually I 'd just engaged a procedure to have the information from my IPST. My former IPST informed me that this kind of device can't be configured for another IPST. Have you already experinced this kind of behaviour? I've never experienced it myself - mostly because I don't buy locked devices. (and that goes for most stuff - e.g. my mobile phone was bought sim-free, no contract, etc.) however I know it goes on - e.g. Vonage are quite famous for it. You might find that googling for your device and ITSP might give you some clues as to hot to unlock it - I'm sure many are unlockable. Do you know some devices that aren't so locked? None of them are locked by default - it's only the service providers that lock them into their own networks - so if you buy anything from an online supplier that doesn't come with any sort of service then you should be fine. Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
Le 15/06/2011 21:14, Gordon Henderson a écrit : On Wed, 15 Jun 2011, Florent THOMAS wrote: Le 15/06/2011 11:06, Florent THOMAS a écrit : Le 15/06/2011 10:53, Gordon Henderson a écrit : On Tue, 14 Jun 2011, Florent THOMAS wrote: Le 12/06/2011 20:41, Florent THOMAS a écrit : Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) I'll try it ASAP and let you know. Hy Gordon, I checked my devices and observed that I'm the owner of 2 LinkSys SPA2102-R configured by keyyo. I used the Voice interaction menu to reset the device. Unfortunately it stills asking me a password. I don't really know much about the SPA devices, sorry. (the only ATAs I've really used have been Grandstreams) It sounds like they might be locked into the keyyo service (if that's an ITSP who provided them pre-configured) You're totally right and actually I 'd just engaged a procedure to have the information from my IPST. My former IPST informed me that this kind of device can't be configured for another IPST. Have you already experinced this kind of behaviour? I've never experienced it myself - mostly because I don't buy locked devices. (and that goes for most stuff - e.g. my mobile phone was bought sim-free, no contract, etc.) however I know it goes on - e.g. Vonage are quite famous for it. You might find that googling for your device and ITSP might give you some clues as to hot to unlock it - I'm sure many are unlockable. Do you know some devices that aren't so locked? None of them are locked by default - it's only the service providers that lock them into their own networks - so if you buy anything from an online supplier that doesn't come with any sort of service then you should be fine. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Gordon, One more time, thank your for answering. My french level of english will appreciate if you can precise your sentence : /You might find that googling for your device and ITSP might give you some clues as to hot to unlock it - _I'm sure many are unlockable. _/ Do you mean that from your point of view most of devices _can't_ be unlocked or the opposite or is there any ironic sense to understand through this sentence? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
On Wed, 15 Jun 2011, Florent THOMAS wrote: Do you know some devices that aren't so locked? None of them are locked by default - it's only the service providers that lock them into their own networks - so if you buy anything from an online supplier that doesn't come with any sort of service then you should be fine. Gordon [snip very hard to follow thread] Linksys devices are locked at the factory AFAIK and cannot be unlocked. If a Linksys ATA is what you are after, you want a model that ends with '-NA'. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
[snip very hard to follow thread] Linksys devices are locked at the factory AFAIK and cannot be unlocked. If a Linksys ATA is what you are after, you want a model that ends with '-NA'. j Thanks for answering. I wasn't looking for a linkSys, I inherit of the device that my customer own for my integration. More generally, I'll looking for an ATA device for plugin Gigaset DECT and other handset to an extension. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
Great, Thanks to all of you for leading me to a solution. regards Le 15/06/2011 21:45, Florent THOMAS a écrit : [snip very hard to follow thread] Linksys devices are locked at the factory AFAIK and cannot be unlocked. If a Linksys ATA is what you are after, you want a model that ends with '-NA'. j Thanks for answering. I wasn't looking for a linkSys, I inherit of the device that my customer own for my integration. More generally, I'll looking for an ATA device for plugin Gigaset DECT and other handset to an extension. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
Hello,
Yes, the issue I am having is currently only with Google Talk. Wonder
if what development will be made to fix this issue.
--Elliot
On Wed, Jun 15, 2011 at 9:20 AM, Vladimir Mikhelson v...@mikhelson.com wrote:
Elliot,
I do not think Issue # 17993 is related. As Terry Wilson says on the
Bug Tracker, Google Voice inbound calls still work, it is just coming
from Google Talk that doesn't.
-Vladimir
On 6/14/2011 5:51 PM, Elliot Murdock wrote:
Hello,
Seems that it's been spotted and tracked at
https://issues.asterisk.org/jira/browse/ASTERISK-17993
--Elliot
On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson v...@mikhelson.com
wrote:
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport
fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk
box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are
the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
--
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Re: [asterisk-users] Google Voice receiving call problem
On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
You should probably grab a free DID as a failover from gtalk. Have gvoice ring them both and answer the one that comes through first. In my tests. I have better luck with the DID than with gtalk. -- cobra2 Http://linuxindixie.info Kevin P. Fleming kpflem...@digium.com wrote: On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dears; OK, I start beleive that the problem in the TFTP and the files that I placed there. Now, I am using the Phone as skinny, and the files that are placed in the directory /var/lib/tftpboot/ as following: CTLSEPB8BEBF22AB62.tlv SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml Well, actually the CTLSEPB8BEBF22AB62.tlv is totally empty, so should I place any thing in it? Anyone has a format for the file CTLSEPB8BEBF22AB62.tlv? Also, what do I miss other files that the Phone needs it? From the other side, what should do about the chown and the chmod for the directory tftpboot? Appreciate the kindly help and advise. Regards Bilal - Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Goggle voice incoming dialplan
Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Web based call back
Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goggle voice incoming dialplan
exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goggle voice incoming dialplan
Thanks and will try. On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue 'ringall' stratagy
Thanks a lot for all your comments. Finally I have figured out the problem by looking into source code. If callcounter=yes and notification is enabled for ringing or hold in sip.conf file, asterisk queue will not fork the new incoming call to the members already in ringing or inuse state. Regards Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com rajib, You can use DIALGROUP function as well On Mon, Jun 13, 2011 at 7:36 PM, Mike l...@net-wall.com wrote: Quite simply: don?t use a queue. Simply ring all phones at the same time using Dial(SIP/phone1SIP/phone2?.) A queue will only send the first call until it is answered, then move on to the second one (I may be simplifying a bit) Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Deka, Rajib IN MAA SL *Sent:* Monday, June 13, 2011 6:44 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] asterisk queue 'ringall' stratagy Hi List, I have faced a problem in asterisk queue implementation. I configured a queue with ?ringall? strategy and ?ringinuse=yes? in queues.conf. If three calls come to this queue in parallel, the logged in queue agent used to get only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed. I want to display all the waiting calls on the agent?s desktop. Is it possible to do, if yes how? Please help me with this. Regards, Rajib *Rajib Deka* SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com -- Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE
Hi Price for Digium (4) span digital T1/E1/J1/PRI PCI card = Rs. 56,000.00 + 5% VAT / 5.00% VAT (Delhi) Price for Digium (4) span digital T1/E1/J1/PRI PCI card with Echo = Rs. 87,000.00 + 5% VAT / 5.00% VAT (Delhi) Price for Sangoma (4) span digital T1/E1/J1/PRI PCI card = Rs. 52,000.00 + 5% VAT /5.00% VAT (Delhi) Warranty – Manufacturer Standard On Wed, Jun 15, 2011 at 4:36 PM, mahesh katta maheshka...@flexydial.comwrote: Hi, I Required digium PRI cards, single span, dual span, quad core . so any body give me cotaion for this cards and I required also grandstream fxs/fxo devices . give me for this quotation . price and details.. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE
Can you provide me express cards price also Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com On Thu, Jun 16, 2011 at 11:02 AM, virendra bhati virbh...@gmail.com wrote: Hi Price for Digium (4) span digital T1/E1/J1/PRI PCI card = Rs. 56,000.00 + 5% VAT / 5.00% VAT (Delhi) Price for Digium (4) span digital T1/E1/J1/PRI PCI card with Echo = Rs. 87,000.00 + 5% VAT / 5.00% VAT (Delhi) Price for Sangoma (4) span digital T1/E1/J1/PRI PCI card = Rs. 52,000.00 + 5% VAT /5.00% VAT (Delhi) Warranty – Manufacturer Standard On Wed, Jun 15, 2011 at 4:36 PM, mahesh katta maheshka...@flexydial.comwrote: Hi, I Required digium PRI cards, single span, dual span, quad core . so any body give me cotaion for this cards and I required also grandstream fxs/fxo devices . give me for this quotation . price and details.. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change destination on digit
Hi, Yes you can use Dial(sip/xxx,30,Ttr) option then it will transfer to any where you want. On Wed, Jun 15, 2011 at 7:03 PM, vip killa vipki...@gmail.com wrote: Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to secure our Asterisk server from hacker's ?
Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
I thought the idea was that Asterisk Engineers already know the answers to such questions? On 06/16/2011 01:52 AM, virendra bhati wrote: Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning via SRV record?
Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com /Proxy_2_ Outbound_Proxy_2_ fqdn /Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Mike. On Tuesday 14 June 2011 3:08:33 am Paul Hayes wrote: On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
