[asterisk-users] Hold
Hi
I am trying to set up a hold system so that a call is always parked in
the same spot no matter how many times it is picked up. My problem is I cannot
fins a variable the identifies the call all the way through until it is
destroyed. ${UNIQUEID} and ${CHANNEL} both seam to get lost when the call is
parked. I tried setting
set($[“${UNIQUEID}-hold”=”701”])
and
set($[“${CHANNEL}-hold”=”701”])
and both work fine until I do a transfer to park. Then both variables are
blank. Is there any variable that is persistent to a call through all of the
transfers.
asterisk 11.6-cert1
centos 5.7
Thanks
Kelly
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[asterisk-users] glibc detected crash
Hi I have asterisk 1.8.18 with freepbx 2.10.1.9. I get an asterisk crash occasionally with the followingerror. It always seems to happen while paging. 16 spa508g phones 1 snom pa1 paging amp Kelly == Extension Changed 1101[ext-paging] new state Idle for Notify User 101 pbx*CLI *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin double linked list corrupted: 0xb6f1dc28 *** *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin double linked list corrupted: 0xb6f1dc28 *** == Extension Changed 104[ext-local] new state Idle for Notify User 106 pbx*CLI === Backtrace: = /lib/libc.so.6(-0xff47c65f)[0x1809a1] /lib/libc.so.6(-0xff478d77)[0x184289] /lib/libc.so.6(-0xff4775cb)[0x185a35] /lib/libc.so.6(realloc+0xdc)[0x185d1c] /usr/sbin/asterisk(ast_event_append_ie_str+0xa6)[0x80ede86] /usr/sbin/asterisk(ast_event_new+0x269)[0x80ee279] /usr/sbin/asterisk[0x80dd038] /usr/sbin/asterisk[0x819807b] /lib/libpthread.so.0[0xcb8a49] /lib/libc.so.6(clone+0x5e)[0x1f263e] === Memory map: 0011-002a r-xp 08:02 2556787/lib/libc-2.12.so 002a-002a2000 r--p 0018f000 08:02 2556787/lib/libc-2.12.so 002a2000-002a3000 rw-p 00191000 08:02 2556787/lib/libc-2.12.so 002a3000-002a6000 rw-p 00:00 0 002a6000-002b r-xp 08:02 722852 /usr/lib/asterisk/modules/pbx_config.so 002b-002b1000 rw-p a000 08:02 722852 /usr/lib/asterisk/modules/pbx_config.so 002b1000-002b4000 r-xp 08:02 723056 /usr/lib/asterisk/modules/res_stun_monitor.so 002b4000-002b5000 rw-p 2000 08:02 723056 /usr/lib/asterisk/modules/res_stun_monitor.so 002b5000-002d2000 r-xp 08:02 723022 /usr/lib/asterisk/modules/res_ael_share.so 002d2000-002d3000 rw-p 0001d000 08:02 723022 /usr/lib/asterisk/modules/res_ael_share.so 002d3000-002d4000 rw-p 00:00 0 002d4000-002e4000 r-xp 08:02 723023 /usr/lib/asterisk/modules/res_agi.so 002e4000-002e6000 rw-p f000 08:02 723023 /usr/lib/asterisk/modules/res_agi.so 002e6000-002f8000 r-xp 08:02 723040 /usr/lib/asterisk/modules/res_fax.so 002f8000-002f9000 rw-p 00011000 08:02 723040 /usr/lib/asterisk/modules/res_fax.so 002f9000-002fb000 r-xp 08:02 723055 /usr/lib/asterisk/modules/res_speech.so 002fb000-002fc000 rw-p 1000 08:02 723055 /usr/lib/asterisk/modules/res_speech.so 002fc000-0030b000 r-xp 08:02 723047 /usr/lib/asterisk/modules/res_odbc.so 0030b000-0030c000 rw-p f000 08:02 723047 /usr/lib/asterisk/modules/res_odbc.so 0030c000-00378000 r-xp 08:02 572483 /usr/lib/libodbc.so.2.0.0 00378000-0037c000 rw-p 0006c000 08:02 572483 /usr/lib/libodbc.so.2.0.0 0037c000-0037d000 rw-p 00:00 0 0037d000-00383000 r-xp 08:02 722855 /usr/lib/asterisk/modules/pbx_lua.so 00383000-00384000 rw-p 5000 08:02 722855 /usr/lib/asterisk/modules/pbx_lua.so 00384000-00388000 r-xp 08:02 723038
[asterisk-users] BLF LED Pattern
Hi We are running incredible PBX (latest version) with 5 spa508g phones. I have 4 of the lines as blf’s to parking spots 71, 72, 73 and 74. The blf goes solid red when someone is parked in that spot. Is there any way to make the blf flash red instead of solid red. Thanks Kelly-- _ -- 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] spa508g and park
Hi I have 16 new cisco(linksys) spa508g phones on asterisk 1.8. I cannot get a on touch park to work. Cisco seems to have the park softkey feature locked to spcp. Has anyone had any luck getting a park button to work. Thanks Kelly-- _ -- 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 and iax2 errors
Hi I am having intermittent iax2 errors on 2 asterisk systems connected with iax2. Below are the errors, and everything I have read online says this was a bug that has been fixed. Both sides are set for trunking with dahdi complete installed. No dahdi hardware just dahdi dummy for timing. Any help would be greatly appreciated. chan_iax2.c: Received trunked frame before first full voice frame server 1 centos 5.6 asterisk 1.8.4.3 dahdi-2.5.0 server 2 centos 5.6 asterisk-1.4.42 dahdi-2.5.0.1 Thanks Kelly-- _ -- 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 config
Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and email
set in astdb. When a tech call comes in the dial plan checks astdb and sets 2
variable ${oc} for the number and ${ocem} for the email. I can easily dial the
number using the variable, but if the call should go to voicemail I am not
getting the email. Below is my voicemail.conf entry for the mailbox. I can hard
code my email and it works so I know there is nothing wrong on the system.
121 = 121,Tech Support,${ocem},1113334...@vtext.net
Any help would be greatly appreciated.
Kelly--
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[asterisk-users] skinny and 7961
Hi I am having a problem with asterisk 1.8.4.2 and chan skinny. I have 10 7961 phones and 2 7920 phones. The 2 7920 phones register without a problem, but the 7961 phones do not. the console message is: chan_skinny.c:6438 get_input: Skinny Client sent less data then expected. The it tries to destroy a nonexistent connection. A have the latest firmware on the phones (9-1-1SR1) any ideas would be greatly appreciated. Kelly-- _ -- 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] dial from voicemail
Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail or to just leave a message at the tone. Thanks Kelly -- _ -- 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] Parking speed
Hi Is it possible to speed up the parking process. We receive a lot of calls during the day and it gets to be painful to wait for the call to be parked and the number to be played back to you. I would like to speed up the whole process including the play back speed. Thanks Kelly___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Park Button
Hi I have a Polycom 501 phone. I set the park feature to 1 in sip.cfg and the button shows up just fine. However when you press it it does nothing. I have the t and T in the dial string. Is there some trick to getting it to work with asterisk 1.4. Thanks Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream 2000 Parking
Hi Is there a way to program #700 to one of the speed dial buttons. I put #700 in the config file for the last button but it's like pressing a dead button. Is there some trick to using the #key to transfer in speed dial buttons. Thanks Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AA50 Paging
Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi It restarts asterisk. The unit does not reboot. Kelly On Mon, 2007-10-15 at 04:22 +0200, Philipp Kempgen wrote: Kelly opal wrote: I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Can't help you with this, but do you mean it reboots/crashes the machine? Or does it restart asterisk? Regards, Philipp Kempgen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi I tried that. Unfortunately it is the Dial command. The first phone to answer wins and the rest are dropped from the channel. Thanks Kelly On Mon, 2007-10-15 at 12:25 +1000, Klaverstyn, David C wrote: I’m not sure if this will work on the Grandstream phones but I use this for the Linksys phones. exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = ,n,Dial(SIP/201) exten = ,n,HangUp I would guess it would work with multiple phones, i.e., exten = ,n,Dial(SIP/201 SIP/202 SIP/203 SIP/204) You may need to check the phone is configured for paging auto answer. The Linksys has a field of Paging Serv and is set to yes. Let me know if it works. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Monday, 15 October 2007 7:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi Digium support says it is built on the 1.4 platform. Kelly On Sun, 2007-10-14 at 22:28 -0400, Joseph Begumisa wrote: Hi, I am curious. What version of asterisk is running on that AA50? Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Sunday, October 14, 2007 5:46 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DSl and more then 1 call
Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated. Kelly ___ --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] DSl and more then 1 call
Hi It's defiantly the branch server. My main server handles 30 to 40 calls at a time on a regular basis. It is only happening on the branch server and it acts like it is using up all the bandwidth of the DSL. It is a 1.5 meg down and 512 up DSL line. I would think it could handle 2 simultaneous calls. I have tried using g729, ulaw, alaw and gsm. There is no difference in the behavior. Could it possible be a routing issue on the LAN side of server 2. Kelly - Original Message - From: Vicky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 13, 2006 1:59 PM Subject: Re: [asterisk-users] DSl and more then 1 call Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server . On 13/11/06, Kelly Opal [EMAIL PROTECTED] wrote: Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated. Kelly___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parking Position
Hi Is there any way to park a call in the same extension all the time. Example: I want Jim at exten 33 to have his calls always parked in exten 703 when he parks a call and Jan at exten 35 to have hers parked in exten 705 when she hits #10. Thanks Kelly Opal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple phones on a Zap FXS extension
Hi I have Asterisk up and running perfect with a Digium TDM400P card and 4 FXS ports. There are 4 ATT 4-Line 954 phones hooked to the system Each of the 4 lines is hooked to each phone. The problem is when you are on line 1 (or any line) and someone else picks up line 1 they can here the conversation. When the phone are hooked to the PSTN in the same way line 1 will be lite up to show that it is in use and you cannot join the call without pressing a certain key sequence. How can I get the phones to act like they do when they are connected to the PSTN when hooked to the Asterisk server. Any help is appreciated. Kelly ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
