Re: [asterisk-users] Asterisk call limitation
check this out: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/258425 From: kche...@xplorium.com To: asterisk-users@lists.digium.com Date: Tue, 21 Jun 2011 13:25:39 +0300 Subject: Re: [asterisk-users] Asterisk call limitation Any update ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Tuesday, June 21, 2011 12:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk call limitation The problem remains even when I add to /etc/init.d/asterisk ulimit -n 65536 [root@localhost ~]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 65536 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 65536 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited [root@localhost ~]# -Original Message- From: Khaled W. Chehab [mailto:kche...@xplorium.com] Sent: Tuesday, June 21, 2011 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk call limitation Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005
Re: [asterisk-users] Asterisk call limitation
It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail:mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com Please refrain from including 20-line signature blocks in your messages to the Asterisk mailing lists (or really, anywhere). Your message had three lines of content and 30+ lines of non-content. -- 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] Asterisk call limitation
Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Monday, June 20, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards
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] Interesting PRI issue
Problem solved. Just changed G1 to g1 -- Sent from my iPhone On Jun 13, 2011, at 9:36 PM, James zhu zhulizh...@live.com wrote: hi: Please check the status of PRI, i think the channels keeps up and down. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/ pri-SIP). website: www.voipviews.com From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 17:44:12 + Subject: [asterisk-users] Interesting PRI issue Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 wanpipe1 card 0 span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 wanpipe2 card 1 span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI == Using SIP RTP CoS mark 5 -- Executing [6463279153@from-sip:1] Dial(SIP/8227-02b1, DAHDI/G1/16463279153) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/16463279153 -- Span 2: Channel 0/23 got hangup, cause 44 -- Span 2: Forcing restart of channel 0/23 since channel reported in use -- Hungup 'DAHDI/i2/16463279153-fe' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/8227-02b1' status is 'CHANUNAVAIL' -- Span 2: Channel 0/23 successfully restarted -- _ -- 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 -- _ -- 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 asterisk use pri channel
I appriciate your reply, But believe me no one option works for me. I tried dahdi/25/XXX but it still using pri first channel or anyother channel In old zap school you can do that but in dahdi I don't think you can. Until unless you create g1 g2 ... Group in chan_dahdi.cfg and map channels there. -- Sent from my iPhone On Jun 9, 2011, at 1:25 AM, Satish Barot satish4aster...@gmail.com wrote: I hope my understanding is not wrong! (1) DAHDI/i2/25/XXX, is not a valid format for Dial. Rather it should be DAHDI/i2/XXX and it would use a channel from span 2 (/etc/dahdi/system.conf) for outgoing call. (2) To dial from channel 25 , use DAHDI/25/XXX [SATISH] On Thu, Jun 9, 2011 at 9:39 AM, satish patel satish...@hotmail.com wrote: Awesome!! Do you know if i want to use only specific channel for call out then how do i write dialplan ? I want to use channel 25 specific for my extension DAHDI/25/ or DAHDI/i2/25/XXX -- _ -- 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] issues.asterisk.org/jira not working
Sure, but how to check which CA my iPhone using ? -- Sent from my iPhone On Jun 8, 2011, at 6:00 PM, Andrew Latham lath...@gmail.com wrote: On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel satish...@hotmail.com wrote: It not working on iPhone. It's saying not able to make secure connection -- Sent from my iPhone Satish, Can you share what the SSL/TLS Cert says? Safari and mobile platforms have a smaller list of CAs, just to make life hard for us sysadmin types... -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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 1.8 broken MWI
Thanks steve, But you know if i connect X-lite softphone my asterisk sending NOTIFY . But its not sending NOTIFY to polycom 501 phone ? Do you think i need to subscribe my phone to asterisk ? -S Date: Wed, 8 Jun 2011 18:15:14 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote: Interesting thing is when i reload sip.conf i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my asterisk not sending NOTIFY message ? Do i need to subscribe my phone to asterisk ? Does this help? https://issues.asterisk.org/jira/browse/ASTERISK-17866 Regards, Steve -- _ -- 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 1.8 broken MWI
Okay! here i have manually configure polycom 501 and tell to subscribe asterisk for MWI. and look like MWI started working but issue is i am getting delayed MWI notification.. sometime its 1 hrs or sometime its 30min see following debug. what is Expires: 3600 ? from where its coming from ? - --- (10 headers 0 lines) --- Really destroying SIP dialog '29bd9ffd4ce2e0b737a68f9145812de2@172.30.1.46:5060' Method: OPTIONS --- SIP read from UDP:172.30.245.143:5060 --- SUBSCRIBE sip:asterisk@172.30.1.46:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.143;branch=z9hG4bK2b7c62c3FA125372 From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE To: sip:7...@laverne.east.ora.com;tag=as65ea68d2 CSeq: 6 SUBSCRIBE Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 Contact: sip:7623@172.30.245.143 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Found peer '7623' for '7623' from 172.30.245.143:5060 Scheduling destruction of SIP dialog '739c15bd-75f452ef-dcd95504@172.30.245.143' in 361 ms (Method: SUBSCRIBE) --- Transmitting (no NAT) to 172.30.245.143:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.143;branch=z9hG4bK2b7c62c3FA125372;received=172.30.245.143 From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE To: sip:7...@laverne.east.ora.com;tag=as65ea68d2 Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 CSeq: 6 SUBSCRIBE Server: Asterisk PBX SVN-branch-1.8-r321926 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: sip:asterisk@172.30.1.46:5060;expires=3600 Content-Length: 0 Reliably Transmitting (no NAT) to 172.30.245.143:5060: NOTIFY sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK337c5799 Max-Forwards: 70 Route: sip:7623@172.30.245.143 From: asterisk sip:asterisk@172.30.1.46;tag=as65ea68d2 To: sip:7623@172.30.245.143;tag=9FBFC6B1-EE9095EE Contact: sip:asterisk@172.30.1.46:5060 Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 CSeq: 107 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 97 Messages-Waiting: yes Message-Account: sip:asterisk@172.30.1.46:5060 Voice-Message: 2/0 (0/0) Date: Thu, 9 Jun 2011 18:25:30 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 9 June 2011 15:49, satish patel satish...@hotmail.com wrote: Date: Wed, 8 Jun 2011 18:15:14 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote: Interesting thing is when i reload sip.conf i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my asterisk not sending NOTIFY message ? Do i need to subscribe my phone to asterisk ? Does this help? https://issues.asterisk.org/jira/browse/ASTERISK-17866 Regards, Steve Thanks steve, But you know if i connect X-lite softphone my asterisk sending NOTIFY . But its not sending NOTIFY to polycom 501 phone ? Do you think i need to subscribe my phone to asterisk ? -S X-Lite automatically SUBSCRIBEs for MWI indication. Polycom and snom do not do this by default, instead they assume that the REGISTER will automatically cause MWI notifications. chan_sip changed behaviour (by accident I suspect) somewhere between version 1.2 and 1.6, and the patch basically puts back what went missing. It is crude, but has not caused me any problems so far. Regards, Steve -- _ -- 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 1.8 broken MWI
How can i change expiry of MWI 003600 last tab campbx1*CLI sip show subscriptions Peer User Call ID ExtensionLast state TypeMailboxExpiry 172.30.245.143 7623 739c15bd-75f452 -- none mwi 7623@defau 003600 172.30.245.143 7623 5e78b9cb-f06bf5 -- none mwi 7623@defau 003600 2 active SIP subscriptions From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jun 2011 17:40:25 + Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Okay! here i have manually configure polycom 501 and tell to subscribe asterisk for MWI. and look like MWI started working but issue is i am getting delayed MWI notification.. sometime its 1 hrs or sometime its 30min see following debug. what is Expires: 3600 ? from where its coming from ? - --- (10 headers 0 lines) --- Really destroying SIP dialog '29bd9ffd4ce2e0b737a68f9145812de2@172.30.1.46:5060' Method: OPTIONS --- SIP read from UDP:172.30.245.143:5060 --- SUBSCRIBE sip:asterisk@172.30.1.46:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.245.143;branch=z9hG4bK2b7c62c3FA125372 From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE To: sip:7...@laverne.east.ora.com;tag=as65ea68d2 CSeq: 6 SUBSCRIBE Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 Contact: sip:7623@172.30.245.143 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Found peer '7623' for '7623' from 172.30.245.143:5060 Scheduling destruction of SIP dialog '739c15bd-75f452ef-dcd95504@172.30.245.143' in 361 ms (Method: SUBSCRIBE) --- Transmitting (no NAT) to 172.30.245.143:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.245.143;branch=z9hG4bK2b7c62c3FA125372;received=172.30.245.143 From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE To: sip:7...@laverne.east.ora.com;tag=as65ea68d2 Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 CSeq: 6 SUBSCRIBE Server: Asterisk PBX SVN-branch-1.8-r321926 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: sip:asterisk@172.30.1.46:5060;expires=3600 Content-Length: 0 Reliably Transmitting (no NAT) to 172.30.245.143:5060: NOTIFY sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK337c5799 Max-Forwards: 70 Route: sip:7623@172.30.245.143 From: asterisk sip:asterisk@172.30.1.46;tag=as65ea68d2 To: sip:7623@172.30.245.143;tag=9FBFC6B1-EE9095EE Contact: sip:asterisk@172.30.1.46:5060 Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143 CSeq: 107 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 97 Messages-Waiting: yes Message-Account: sip:asterisk@172.30.1.46:5060 Voice-Message: 2/0 (0/0) Date: Thu, 9 Jun 2011 18:25:30 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 9 June 2011 15:49, satish patel satish...@hotmail.com wrote: Date: Wed, 8 Jun 2011 18:15:14 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote: Interesting thing is when i reload sip.conf i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my asterisk not sending NOTIFY message ? Do i need to subscribe my phone to asterisk ? Does this help? https://issues.asterisk.org/jira/browse/ASTERISK-17866 Regards, Steve Thanks steve, But you know if i connect X-lite softphone my asterisk sending NOTIFY . But its not sending NOTIFY to polycom 501 phone ? Do you think i need to subscribe my phone to asterisk ? -S X-Lite automatically SUBSCRIBEs for MWI indication. Polycom and snom do not do this by default, instead they assume that the REGISTER will automatically cause MWI notifications. chan_sip changed behaviour (by accident I suspect) somewhere between version 1.2 and 1.6, and the patch basically puts back what went missing. It is crude, but has not caused me any problems so far. Regards, Steve -- _ -- 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
[asterisk-users] Polycom 501 Settings/subscription expiry
Hi, Anybody know how to set polycom 501 subscription expiry ? -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] PRI hangup request, cause 18
Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle supp...@drdos.info wrote: satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] PRI hangup request, cause 18
We have two sites. BOSTON and California We are having only issue with California PRI line related cause 18 but BOSTON pri has no issue. All settings are same on both Asterisk. Today i will talk to service provider and will see. pridialplan=uknown fixed many issues except cause 18 -S Date: Wed, 8 Jun 2011 15:41:04 +0200 From: t...@ovm-group.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI hangup request, cause 18 Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination - for whatever reason. Am 08.06.2011 12:55, schrieb Satish Patel: Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle supp...@drdos.info wrote: satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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 1.8 broken MWI
Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 1.8 broken MWI
Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 1.8 broken MWI
Following is my debug and look like its not sending MWI NOTIFY message to phone Reliably Transmitting (no NAT) to 172.30.245.143:5060: OPTIONS sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3 Max-Forwards: 70 From: asterisk sip:asterisk@172.30.1.46;tag=as26352734 To: sip:7623@172.30.245.143 Contact: sip:asterisk@172.30.1.46:5060 Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Date: Wed, 08 Jun 2011 14:49:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:172.30.245.143:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3 From: asterisk sip:asterisk@172.30.1.46;tag=as26352734 To: sip:7623@172.30.245.143;tag=E777D3B9-F605D562 CSeq: 102 OPTIONS Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060 Contact: sip:7623@172.30.245.143 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 - --- (10 headers 0 lines) --- Really destroying SIP dialog '44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 172.30.245.143:5060: OPTIONS sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37 Max-Forwards: 70 From: asterisk sip:asterisk@172.30.1.46;tag=as0c8778f4 To: sip:7623@172.30.245.143 Contact: sip:asterisk@172.30.1.46:5060 Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Date: Wed, 08 Jun 2011 14:50:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:172.30.245.143:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37 From: asterisk sip:asterisk@172.30.1.46;tag=as0c8778f4 To: sip:7623@172.30.245.143;tag=47557FCE-869CEA2F CSeq: 102 OPTIONS Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060 Contact: sip:7623@172.30.245.143 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 - --- (10 headers 0 lines) --- From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 14:43:57 + Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 -- _ -- 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 1.8 broken MWI
I do have that sip.conf [7623](cam-exten) callerid=Satish Patel 7623 accountcode=Satish Patel mailbox=7623@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:03:24 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 10:44 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 -- _ -- 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 1.8 broken MWI
Do you think i should enable ? ; searchcontexts=yes From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:03:24 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 10:44 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 -- _ -- 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 1.8 broken MWI
Yes its under [defailt] section at voicemail.conf From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:17:26 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Is 7623 listed in voicemail.conf under the [default] section? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 11:15 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI I do have that sip.conf [7623](cam-exten) callerid=Satish Patel 7623 accountcode=Satish Patel mailbox=7623@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:03:24 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 10:44 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 -- _ -- 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 1.8 broken MWI
Yes its under [defailt] section at voicemail.conf Sorry it my typo error. When there is a new message in a mailbox, does voicemail show users show new messages for that mailbox? Yes, I can see there are 10 voicemail root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623 default7623 Satish Patel 10 From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:33:31 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI I assume you misspelled default in your e-mail and not voicemail.conf. If not, that is your problem. When there is a new message in a mailbox, does voicemail show users show new messages for that mailbox? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 11:21 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Yes its under [defailt] section at voicemail.conf From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:17:26 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Is 7623 listed in voicemail.conf under the [default] section? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 11:15 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI I do have that sip.conf [7623](cam-exten) callerid=Satish Patel 7623 accountcode=Satish Patel mailbox=7623@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:03:24 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 10:44 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk 1.8 broken MWI
Interesting thing is when i reload sip.conf i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my asterisk not sending NOTIFY message ? Do i need to subscribe my phone to asterisk ? From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 15:38:53 + Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Yes its under [defailt] section at voicemail.conf Sorry it my typo error. When there is a new message in a mailbox, does voicemail show users show new messages for that mailbox? Yes, I can see there are 10 voicemail root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623 default7623 Satish Patel 10 From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:33:31 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI I assume you misspelled default in your e-mail and not voicemail.conf. If not, that is your problem. When there is a new message in a mailbox, does voicemail show users show new messages for that mailbox? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 11:21 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Yes its under [defailt] section at voicemail.conf From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:17:26 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Is 7623 listed in voicemail.conf under the [default] section? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 11:15 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI I do have that sip.conf [7623](cam-exten) callerid=Satish Patel 7623 accountcode=Satish Patel mailbox=7623@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 11:03:24 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 10:44 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 10:34:16 -0400 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 9:57 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 broken MWI Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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] Interesting PRI issue
Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 wanpipe1 card 0 span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 wanpipe2 card 1 span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI == Using SIP RTP CoS mark 5 -- Executing [6463279153@from-sip:1] Dial(SIP/8227-02b1, DAHDI/G1/16463279153) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/16463279153 -- Span 2: Channel 0/23 got hangup, cause 44 -- Span 2: Forcing restart of channel 0/23 since channel reported in use -- Hungup 'DAHDI/i2/16463279153-fe' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/8227-02b1' status is 'CHANUNAVAIL' -- Span 2: Channel 0/23 successfully restarted -- _ -- 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] issues.asterisk.org/jira not working
Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- 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 asterisk use pri channel
Hi, We have two pri line and I want to see how asterisk distribute outgoing call per channels I meant it use first last channel 47 or it will use first channel? Or it will allocate dynamically ? -- Sent from my iPhone -- _ -- 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 asterisk use pri channel
Awesome!! Do you know if i want to use only specific channel for call out then how do i write dialplan ? I want to use channel 25 specific for my extension DAHDI/25/ or DAHDI/i2/25/XXX Date: Wed, 8 Jun 2011 17:25:44 -0500 From: rmudg...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How asterisk use pri channel We have two pri line and I want to see how asterisk distribute outgoing call per channels I meant it use first last channel 47 or it will use first channel? Or it will allocate dynamically ? Extracted from chan_dahdi.c: Dial(DAHDI/pseudo[/extension[/options]]) Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]]) Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]]) Dial(DAHDI/ispan[/extension[/options]]) Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]]) i - ISDN span channel restriction. Used by CC to ensure that the CC recall goes out the same span. Also to make ISDN channel names dialable when the sequence number is stripped off. (Used by DTMF attended transfer feature.) g - channel group allocation search forward G - channel group allocation search backward r - channel group allocation round robin search forward R - channel group allocation round robin search backward c - Wait for DTMF digit to confirm answer rcadance# - Set distintive ring cadance number d - Force bearer capability for ISDN/SS7 call to digital. All are valid for v1.8 and trunk. The ispan option and subdir! option are not valid earlier than v1.8. Richard -- _ -- 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] reload chan_dahdi.conf without disconnect active calls
Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -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
[asterisk-users] reload chan_dahdi.conf without disconnect active calls
Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -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
[asterisk-users] PRI hangup request, cause 18
We have 2 PRI from ATT And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause, and clearing call [Jun 7 17:57:33] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/4 got hangup request, cause 18 -- _ -- 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 1.8 issue with polycom dialplan
Hi all, I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from _71XX. now what happen if i dial any 711X number my polycom just dial 711 and say busy number look like my phone doing some regex itself. like 911 number.. Did you get what i am trying to say ? it was working before with 1.2 but after upgrade 1.8 it started issue. why its just going with 711* 611* 511* etc... -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] asterisk 1.8 issue with polycom dialplan
look like we found issue in phone configuration files [2-9]xx From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:43:22 + Subject: [asterisk-users] asterisk 1.8 issue with polycom dialplan Hi all, I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from _71XX. now what happen if i dial any 711X number my polycom just dial 711 and say busy number look like my phone doing some regex itself. like 911 number.. Did you get what i am trying to say ? it was working before with 1.2 but after upgrade 1.8 it started issue. why its just going with 711* 611* 511* etc... -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 -- _ -- 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] PRI issue its BUSY
sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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] PRI issue its BUSY
This is wired.. If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound calls.. But its not working with asterisk 1.8 :( ( i can call in but not out) From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:11:28 + Subject: Re: [asterisk-users] PRI issue its BUSY sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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 -- _ -- 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] [SOLVED]PRI issue its BUSY
Solution: pridialplan=unknow From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:33:44 + Subject: Re: [asterisk-users] PRI issue its BUSY This is wired.. If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound calls.. But its not working with asterisk 1.8 :( ( i can call in but not out) From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:11:28 + Subject: Re: [asterisk-users] PRI issue its BUSY sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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 -- _ -- 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] broken SVN asterisk 1.8 ?
Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 -- _ -- 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] broken SVN asterisk 1.8 ?
Thanks but they should change svn revesion number change in file. -- Sent from my iPhone On Jun 5, 2011, at 7:13 PM, Barry Miller asterisk-us...@notanet.net wrote: On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote: Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 asterisk -V shows the last changed revision in the build. To see the difference, try: cd asterisk-src-dir svnversion svnversion -c -- Barry -- _ -- 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] benefits of asterisk 1.8
Yesterday my 1.8 got crashed and I have nothing in log or anywhere which I can show you or submit bug. Kinda funny :( -- Sent from my iPhone On Jun 3, 2011, at 5:06 AM, Satish Barot satish4aster...@gmail.com wrote: If 1.8 doesn't panic for subset of PBX features for someone, you can not say it is stable. You should also look at other features and how they work with 1.8. I didn't say 1.4 or 1.6 have no bugs or issues. When there were 1.4 or 1.6.0 branches, they did have bugs. But since people started submitting bug reports, they have become quite stable. They don't get crashed as frequently as 1.8 for the same set of features(You can check it on issues.asterisk.org). When I said 'Asterisk 1.8 is not stable ENOUGH', I didn't mean 'Asterisk 1.8 is not stable AT ALL'.There are still some feature functionalities which work perfactaly on 1.4 or 1.6, create some panic on 1.8. I would consider 1.8 stable enough when anything which worked on 1.4 or 1.6, also work on 1.8. And I am optimistic about 1.8 being stable enough shortly. Let us not start a war on 1.8 stability issue. There were enough threads on 1.8 being production safe in last couple of months.Mine was just a user experience and personal view shared with somebody else. [SATISH] On Fri, Jun 3, 2011 at 1:37 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' Ish __ -- 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 -- _ -- 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] Queue base polycom custom ringtype
Hey Guy, I want to implement Queue base custom ring tone so Agent will get aware of incoming call for sale or tech etc.. I know its possible with SIPAddHeader http://www.technicallyamusing.com/?p=44 I am confused here alertInfo voIpProt.SIP.alertInfo.1.value=custome-ring voIpProt.SIP.alertInfo.1.class=5 We already have alertInfo set to Ring Answer how should i use both ring and Ring Answer ? alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ -- _ -- 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] benefits of asterisk 1.8
Sherwood, I was wrong here But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? I have just cross check and we have option OPTIMIZED. That mean don't create coredump right ? -S Date: Fri, 3 Jun 2011 09:53:01 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 Message body On 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- 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 1.8.4.2 Now Available (Security Release)
Is this available in current SVN ? Date: Thu, 2 Jun 2011 15:07:50 -0400 From: asteriskt...@digium.com To: asteriskt...@digium.com Subject: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release) The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash: Remote Crash Vulnerability in SIP channel driver (AST-2011-007) The issue and resolution is described in the AST-2011-007 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-007, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2 Security advisory AST-2011-007 is available at: http://downloads.asterisk.org/pub/security/AST-2011-007.pdf Thank you for your continued support of Asterisk! -- _ -- 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 logger permission
Hi Guys! If i reload my asterisk it create /var/log/asterisk/* file with root permission. I am running asterisk with asterisk user and group. Do you have any idea ? root@campbx1:~# ls -l /var/log/asterisk/ total 716 drwxr-xr-x 2 asterisk asterisk 4096 2011-05-06 15:38 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cdr-custom drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cel-csv drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cel-custom -rw-r- 1 root root 0 2011-05-15 06:25 full -rw-r- 1 asterisk asterisk 617026 2011-05-15 06:25 full.1 -rw-r--r-- 1 asterisk asterisk 41439 2011-05-08 11:24 full.2.gz -rw-r- 1 root root 0 2011-05-15 06:25 messages -rw-r- 1 asterisk asterisk 36519 2011-05-14 19:29 messages.1 -rw-r--r-- 1 asterisk asterisk 2520 2011-05-06 17:21 messages.2.gz -rw-r- 1 root root 0 2011-05-15 06:25 queue_log -rw-r--r-- 1 asterisk asterisk392 2011-05-12 17:23 queue_log.1 -- _ -- 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 fails when DNS or internet fails
I our setup we don't have DNS or Internet connectivity but we are good no issue so far. -- Sent from my iPhone On May 31, 2011, at 7:24 AM, Hans Witvliet h...@a-domani.nl wrote: On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote: On Mon, 30 May 2011, Sherwood McGowan wrote: True, but with all due respect, if the cache's TTL expires and the OP's PBX cannot reach an external DNS server, they have bigger problems ;-) Slainte all! The Mick I couldn't disagree more. In fact I think this problem is more serious than it is getting credit for, when asterisk is in use in places where Internet connectivity is far from stable. I have several hotels that have gone without Internet connectivity for days, and somewhere between one and three days down they can only spottily call within the system, and can't make outbound calls on their voice T1. Its certainly true that they were suffering without Internet access, but it is very hard to explain to the owners why they can't use their phones. In fact the symptoms are very strange - inbound calls on the T1 get the auto-attendant, but internal transfers fail. No one can call outbound, and only *sometimes* do internal extension to extension calls fail. I still scratch my head about what exactly asterisk is trying to lookup that keeps it from being able to place internal SIP calls from extension to extension, and sadly the few times this has occurred I wasn't around to debug. Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? What kind of info is it about? If it is the hostname of _local_ machines/clients, you should be authoritive. That should keep asterisk happy. If it is about remote nodes, well if your isp-connection is lost, you can not contact them anyway ;-( So run locally your bind-server, authoritive for your own addresses, and caching for external ones. hw -- _ -- 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] queuemetrics with 1.8 queue_log
Hi Guys! We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/ instead of Agent/ that is obvious behaviors. so do i need to change Agent/ to SIP/ in queuemetrics ? or is there any workaround to keep business running same like it was before. -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
[asterisk-users] Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)
Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -- _ -- 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] please help
Did you try different number in place of 5? I meant 1 2 etc..
Also check cli logs on console
Are you dialing from softphone or hardphone because some phone has
dialing regex for security.
--
Sent from my iPhone
On May 30, 2011, at 1:30 PM, salaheddine elharit salah.elharit...@gmail.com
wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av
(0}V(0))
exten = _0678922645.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-
monitored-or-recorded))
exten = _0678922645,2,Hangup()
i can not call my number but when i delet the last number '5' i can
call without any issue
i want to put all the number please any hel to solve this issue
thanks and regards
--
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Re: [asterisk-users] Asterisk 1..8 multiple queue
That's cool. I will give it a shot and let you guys know. -- Sent from my iPhone On May 27, 2011, at 5:18 AM, Paul Hayes p...@provu.co.uk wrote: On 26/05/11 23:18, Satish Patel wrote: Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? From the example: *CLI database put queue_agent 0001/available_queues support^sales support^sales is a list of queues. Put as many in the list as you need. E.G. sales^support^tech 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 -- _ -- 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] DID for outbound PSTN call
Hi There, We have single PRI with multiple DID numbers and its working fine in receiving call. And if you make outbound call it will send main-line CallerID (company name). Now we want individual caller id for per extensions on outbound calls. like if i call someone he will get my extension as callerid ( 617-838-) is my sip extension something like this so next time i direct get call from users. How to do this ? -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] DID for outbound PSTN call
That is very cool,
Is that means it will overwrite my global callerid setting at dahdi-channels?
root@sfpbx1:/home/satish# cat /etc/asterisk/dahdi-channels.conf | grep callerid
callerid=6178387100
-S
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
Date: Fri, 27 May 2011 10:45:32 -0400
Subject: Re: [asterisk-users] DID for outbound PSTN call
Add Set(CALLERID(num)=617838${CALLERID(num)}) to your dialplan for outgoing
calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Friday, May 27, 2011 10:42 AM
To: asterisk-users
Subject: [asterisk-users] DID for outbound PSTN call
Hi There,
We have single PRI with multiple DID numbers and its working
fine in receiving call. And if you make outbound call it will
send main-line CallerID (company name). Now we want
individual caller id for per extensions on outbound calls.
like if i call someone he will get my extension as callerid
( 617-838-) is my sip extension something like this
so next time i direct get call from users. How to do this ?
-S
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[asterisk-users] DAHDI span timeing source
Hi There, We have very old asterisk 1.2 running in production and it has following setting in /etc/zaptel.conf. I have read on web about span and they told span= span num ,timing source,line build out (LBO),framing,coding[,yellow] Just wondering why it has timing source 0 ? 0=master, 1=slave right ? Do you think i should change it to 1 ? #Sangoma A102 port 1 [slot:2 bus:7 span:1] wanpipe1 span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #Sangoma A102 port 2 [slot:2 bus:7 span:2] wanpipe2 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 -- _ -- 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 1..8 multiple queue
This is working great! Thanks a lot paul. One more question before we have Agent/ configured in queueMetrics so i need to change them in queueMetrics with SIP/ right ? Date: Fri, 27 May 2011 10:18:39 +0100 From: p...@provu.co.uk To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue On 26/05/11 23:18, Satish Patel wrote: Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? From the example: *CLI database put queue_agent 0001/available_queues support^sales support^sales is a list of queues. Put as many in the list as you need. E.G. sales^support^tech 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 -- _ -- 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 1..8 multiple queue
Oh! wait i got following error when i trying to Unpause my queue. do you have any idea ? holler*CLI == Using SIP RTP CoS mark 5 -- Executing [*99@from-sip:1] Verbose(SIP/7102-000e, 2,UnPausing member in all queues) in new stack == UnPausing member in all queues -- Executing [*99@from-sip:2] Gosub(SIP/7102-000e, subSetupAvailableQueues,start,1()) in new stack -- Executing [start@subSetupAvailableQueues:1] Verbose(SIP/7102-000e, 2,Checking for available queues) in new stack == Checking for available queues -- Executing [start@subSetupAvailableQueues:2] Set(SIP/7102-000e, MemberChannel=7102) in new stack -- Executing [start@subSetupAvailableQueues:3] Set(SIP/7102-000e, MemberChanType=SIP) in new stack -- Executing [start@subSetupAvailableQueues:4] Set(SIP/7102-000e, AvailableQueues=booktech1^booktech2) in new stack -- Executing [start@subSetupAvailableQueues:5] GotoIf(SIP/7102-000e, 0?no_queues_available,1) in new stack -- Executing [start@subSetupAvailableQueues:6] Return(SIP/7102-000e, ) in new stack -- Executing [*99@from-sip:3] UnpauseQueueMember(SIP/7102-000e, ,SIP/7102) in new stack [May 27 11:40:19] WARNING[2358]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = PAUSED ^ [May 27 11:40:19] WARNING[2358]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables -- Executing [*99@from-sip:4] GotoIf(SIP/7102-000e, ?agent_unpaused,1:agent_not_found,1) in new stack -- Goto (from-sip,agent_not_found,1) -- Executing [agent_not_found@from-sip:1] Verbose(SIP/7102-000e, 2,Agent was not found) in new stack == Agent was not found -- Executing [agent_not_found@from-sip:2] Playback(SIP/7102-000e, silence/1cannot-complete-as-dialed) in new stack -- SIP/7102-000e Playing 'silence/1.ulaw' (language 'en') -- SIP/7102-000e Playing 'cannot-complete-as-dialed.ulaw' (language 'en') -- Auto fallthrough, channel 'SIP/7102-000e' status is 'UNKNOWN' From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 27 May 2011 18:03:02 + Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue This is working great! Thanks a lot paul. One more question before we have Agent/ configured in queueMetrics so i need to change them in queueMetrics with SIP/ right ? Date: Fri, 27 May 2011 10:18:39 +0100 From: p...@provu.co.uk To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue On 26/05/11 23:18, Satish Patel wrote: Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? From the example: *CLI database put queue_agent 0001/available_queues support^sales support^sales is a list of queues. Put as many in the list as you need. E.G. sales^support^tech 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 -- _ -- 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] [SOLVED] Asterisk 1..8 multiple queue
In this book example there is a printing issue at Unpaused section. it should
be like following
same = n,GotoIf($[${UPQMSTATUS} = UNPAUSED]?agent_unpaused,1:agent_not_found,1)
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 27 May 2011 18:41:18 +
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
Oh! wait i got following error when i trying to Unpause my queue. do you have
any idea ?
holler*CLI
== Using SIP RTP CoS mark 5
-- Executing [*99@from-sip:1] Verbose(SIP/7102-000e, 2,UnPausing
member in all queues) in new stack
== UnPausing member in all queues
-- Executing [*99@from-sip:2] Gosub(SIP/7102-000e,
subSetupAvailableQueues,start,1()) in new stack
-- Executing [start@subSetupAvailableQueues:1] Verbose(SIP/7102-000e,
2,Checking for available queues) in new stack
== Checking for available queues
-- Executing [start@subSetupAvailableQueues:2] Set(SIP/7102-000e,
MemberChannel=7102) in new stack
-- Executing [start@subSetupAvailableQueues:3] Set(SIP/7102-000e,
MemberChanType=SIP) in new stack
-- Executing [start@subSetupAvailableQueues:4] Set(SIP/7102-000e,
AvailableQueues=booktech1^booktech2) in new stack
-- Executing [start@subSetupAvailableQueues:5] GotoIf(SIP/7102-000e,
0?no_queues_available,1) in new stack
-- Executing [start@subSetupAvailableQueues:6] Return(SIP/7102-000e,
) in new stack
-- Executing [*99@from-sip:3] UnpauseQueueMember(SIP/7102-000e,
,SIP/7102) in new stack
[May 27 11:40:19] WARNING[2358]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected '=', expecting $end; Input:
= PAUSED
^
[May 27 11:40:19] WARNING[2358]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [*99@from-sip:4] GotoIf(SIP/7102-000e,
?agent_unpaused,1:agent_not_found,1) in new stack
-- Goto (from-sip,agent_not_found,1)
-- Executing [agent_not_found@from-sip:1] Verbose(SIP/7102-000e,
2,Agent was not found) in new stack
== Agent was not found
-- Executing [agent_not_found@from-sip:2] Playback(SIP/7102-000e,
silence/1cannot-complete-as-dialed) in new stack
-- SIP/7102-000e Playing 'silence/1.ulaw' (language 'en')
-- SIP/7102-000e Playing 'cannot-complete-as-dialed.ulaw' (language
'en')
-- Auto fallthrough, channel 'SIP/7102-000e' status is 'UNKNOWN'
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 27 May 2011 18:03:02 +
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
This is working great! Thanks a lot paul.
One more question before we have Agent/ configured in queueMetrics so i
need to change them in queueMetrics with SIP/ right ?
Date: Fri, 27 May 2011 10:18:39 +0100
From: p...@provu.co.uk
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
On 26/05/11 23:18, Satish Patel wrote:
Thanks,
I went through this example before. I was confuse and wondering how
should I add third queue in this picture?
From the example:
*CLI database put queue_agent 0001/available_queues support^sales
support^sales is a list of queues. Put as many in the list as you
need. E.G. sales^support^tech
cheers,
Paul.
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Re: [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue
This has been submitted.
-S
Date: Fri, 27 May 2011 16:05:28 -0400
From: leif.mad...@asteriskdocs.org
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue
On 27/05/11 03:18 PM, satish patel wrote:
In this book example there is a printing issue at Unpaused section. it
should be like following
same = n,GotoIf($[${UPQMSTATUS} =
UNPAUSED]?agent_unpaused,1:agent_not_found,1)
Please file stuff like this as errata at
http://oreilly.com/catalog/9780596517342 (left hand side). That way we
can get it fixed up in subversion.
Thanks!
Leif.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] DAHDI span timeing source
You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Date: Fri, 27 May 2011 17:27:43 -0300 From: rafaels...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DAHDI span timeing source Hi The timing source is the clock of the system. When a equipment is 0, the other should be 1. The correct is: 0=slave, 1=master. The default for private systems is slave. Att,Rafael Saraiva 2011/5/27 satish patel satish...@hotmail.com Hi There, We have very old asterisk 1.2 running in production and it has following setting in /etc/zaptel.conf. I have read on web about span and they told span= span num ,timing source,line build out (LBO),framing,coding[,yellow] Just wondering why it has timing source 0 ? 0=master, 1=slave right ? Do you think i should change it to 1 ? #Sangoma A102 port 1 [slot:2 bus:7 span:1] wanpipe1 span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #Sangoma A102 port 2 [slot:2 bus:7 span:2] wanpipe2 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 -- _ -- 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 -- Att,Rafael Saraiva -- _ -- 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] DAHDI span timeing source
Tell me in one word. We have 2 PRI line connected with sangoma card what option would be good for me? 0 or 1 ? -S Date: Fri, 27 May 2011 16:11:03 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DAHDI span timeing source On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote: You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Not really. Looking in system.conf.sample in dahdi-tools [1] Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred source of the master clock. Choose 2 to make it the second choice for the master clock, if the first choice port fails (the far end dies, a cable breaks, or whatever). Choose 3 to make a port the third choice, and so on. If you have, say, 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each port should be different. If you choose 0, the port will never be used as a source of timing. This is appropriate when you know the far end should always be a slave to you. If the port is connected to a channel bank, for example, you should always be its master. Likewise, BRI TE ports should always be configured as a slave. Any number of ports can be marked as 0. [1] http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] DAHDI span timeing source
Got it but still confused. As per your example I should go with Port 1 Span=1,1,0 Port 2 Span=2,2,0 Correct me if I'm wrong. -- Sent from my iPhone On May 27, 2011, at 5:32 PM, Shaun Ruffell sruff...@digium.com wrote: On Fri, May 27, 2011 at 09:20:46PM +, satish patel wrote: Tell me in one word. We have 2 PRI line connected with sangoma card what option would be good for me? 0 or 1 ? Look at the two last sentences of the first paragraph I quoted below. I believe that is your answer...and it's not 0 or 1. Date: Fri, 27 May 2011 16:11:03 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DAHDI span timeing source On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote: You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Not really. Looking in system.conf.sample in dahdi-tools [1] Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred source of the master clock. Choose 2 to make it the second choice for the master clock, if the first choice port fails (the far end dies, a cable breaks, or whatever). Choose 3 to make a port the third choice, and so on. If you have, say, 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each port should be different. If you choose 0, the port will never be used as a source of timing. This is appropriate when you know the far end should always be a slave to you. If the port is connected to a channel bank, for example, you should always be its master. Likewise, BRI TE ports should always be configured as a slave. Any number of ports can be marked as 0. [1] http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] DAHDI span timeing source
It's connected to teclo ATT PSTN for outside calling. So definitly they are master and we are slave but I'm confused about 0 is master or slave? Because few people saying 1 is master and 0 is slave ? I didn't find any clear document every one trying to explain science but none of clear. -- Sent from my iPhone On May 27, 2011, at 5:41 PM, Edwin Lam edwin@officegeneral.com wrote: On 5/27/11 2:20 PM, satish patel wrote: Tell me in one word. We have 2 PRI line connected with sangoma card what option would be good for me? 0 or 1 ? that would depends on what's the other end of the 2 PRI connected to. Date: Fri, 27 May 2011 16:11:03 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DAHDI span timeing source On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote: You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Not really. Looking in system.conf.sample in dahdi-tools [1] Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred source of the master clock. Choose 2 to make it the second choice for the master clock, if the first choice port fails (the far end dies, a cable breaks, or whatever). Choose 3 to make a port the third choice, and so on. If you have, say, 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each port should be different. If you choose 0, the port will never be used as a source of timing. This is appropriate when you know the far end should always be a slave to you. If the port is connected to a channel bank, for example, you should always be its master. Likewise, BRI TE ports should always be configured as a slave. Any number of ports can be marked as 0. [1] http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] DAHDI span timeing source
Thanks also let me clear one thing this pri is PSTN connected to ATT techo. So they are master. -- Sent from my iPhone On May 27, 2011, at 5:51 PM, Shaun Ruffell sruff...@digium.com wrote: On Fri, May 27, 2011 at 05:40:30PM -0400, Satish Patel wrote: Got it but still confused. As per your example I should go with Port 1 Span=1,1,0 Port 2 Span=2,2,0 Correct me if I'm wrong. Yes. That looks correct based on my understanding of your situation. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] DAHDI span timeing source
I guess you are wrong here correct one is 0=master 1=slave If you connect to PSTN the you should user span=1,1,0 Check out http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html -- Sent from my iPhone On May 27, 2011, at 4:27 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi The timing source is the clock of the system. When a equipment is 0, the other should be 1. The correct is: 0=slave, 1=master. The default for private systems is slave. Att, Rafael Saraiva 2011/5/27 satish patel satish...@hotmail.com Hi There, We have very old asterisk 1.2 running in production and it has following setting in /etc/zaptel.conf. I have read on web about span and they told span= span num ,timing source,line build out (LBO),framing,coding[,yellow] Just wondering why it has timing source 0 ? 0=master, 1=slave right ? Do you think i should change it to 1 ? #Sangoma A102 port 1 [slot:2 bus:7 span:1] wanpipe1 span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #Sangoma A102 port 2 [slot:2 bus:7 span:2] wanpipe2 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 -- _ -- 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 -- Att, Rafael Saraiva -- _ -- 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 1..8 multiple queue
Actually right now i have very big AddQueueMember dialplan for every individual queue for login/logout/pause/unpause etc.. ( we have 3 queue) Let me explain my example We have 3 queues ( sales, support, tech) Sales - A,B,C,D,E agents Support - A,B,C,D,E agents tech - A,Z agents Before it was quite simple just specify member in queue but with AddQueueMember its now that case. Before it was just single queue login allowed you to enter in all queue. but in AddQueueMember they have very complex agent login thing. Could you give me example or tell me how i use AddQueueMember in my current setup which i explain you. (multiple queue login and restrict agent for other queue) -S CC: asterisk-users@lists.digium.com From: sherwood.mcgo...@gmail.com Date: Wed, 25 May 2011 20:59:06 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue Yes, there are other ways, I was only offering the solution that has worked best for me. Keep in mind, you are not limited to MySQL for realtime, Asterisk can use any ODBC DSN for the data backend. Oracle, Access, MSSQL are all examples, if I recall correctly you can even connect SQLite and DB2. However, let me ask you this...what trouble are you having with AddQueueMember and it's related applications that is making it hard for you? Sent from my iPhone On May 25, 2011, at 7:20 PM, Satish Patel satish...@hotmail.com wrote: Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. -- Sent from my iPhone On May 25, 2011, at 6:43 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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 -- _ -- 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 1..8 multiple queue
Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? -- Sent from my iPhone On May 26, 2011, at 5:43 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 26/05/11 04:20 PM, satish patel wrote: Actually right now i have very big AddQueueMember dialplan for every individual queue for login/logout/pause/unpause etc.. ( we have 3 queue) Let me explain my example We have 3 queues ( sales, support, tech) Sales - A,B,C,D,E agents Support - A,B,C,D,E agents tech - A,Z agents Before it was quite simple just specify member in queue but with AddQueueMember its now that case. Before it was just single queue login allowed you to enter in all queue. but in AddQueueMember they have very complex agent login thing. Could you give me example or tell me how i use AddQueueMember in my current setup which i explain you. (multiple queue login and restrict agent for other queue) The solution to your problem is to write some dialplan. I even helped you along by writing some documentation :) http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html#ACD_id288626 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 1..8 multiple queue
Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -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] Asterisk 1..8 multiple queue
Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. -- Sent from my iPhone On May 25, 2011, at 6:43 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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] Agent (Invalid) has taken no calls yet
Great! Satish, I am middle of migration 1.2 queue in 1.8 thats why i encounter there. if i add SIP/XXX then my queue working fine. Also i don't understand relation between agents.conf and member = at queues.conf let me read that URL and see what i can find there. -S Date: Fri, 20 May 2011 09:58:59 +0530 From: satish4aster...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Agent (Invalid) has taken no calls yet If you go for 1.8,Don't read from http://www.asteriskguru.com/tutorials/queues.html. It is bit backdated information. Rather I would suggest you to check http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html. Queue members are considered INVALID, if their device status is Invalid. This is somewhat an error condition.SIP channels are the only type that provide true device state information. I also suggest you to read 'The agents.conf File' section from given link for more information. [SATISH] On Fri, May 20, 2011 at 2:40 AM, satish patel satish...@hotmail.com wrote: How to get rid on following.. why its Invalid ? holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -- _ -- 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 -- _ -- 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] Static agent in queue
Hi, I want to add static agent in queue so how to do that it seem 1.8 has very different approach. I have added SIP extension but they are not getting calls. @queues.conf member = SIP/blah member = SIP/blah -- _ -- 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] Agent (Invalid) has taken no calls yet
I do have agents in agents.conf. I am not using agentlogin apps. I am using AddQueueMember agent = 7101,,Agent1 agent = 7102,,Agent2 From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Fri, 20 May 2011 11:56:23 -0500 Subject: Re: [asterisk-users] Agent (Invalid) has taken no calls yet On Thu, 2011-05-19 at 21:10 +, satish patel wrote: How to get rid on following.. why its Invalid ? holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers Your agents are invalid because they are not pointing to a valid device. Is the agent defined in agents.conf? When the agent logs in is he/she passing the correct extension to agentlogin? Maybe it is time to consider dynamic agents for your queues? Since agentcallbacklogin was deprecated in 1.6 I think static agents are more of a bother than they are worth. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] Restart asterisk destroy all registered SIP peers
Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks 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] Restart asterisk destroy all registered SIP peers
We have polycom 501 and i am waiting since last 5 min no registration require appear. -S From: mden...@gmail.com Date: Fri, 20 May 2011 14:56:20 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers On Fri, May 20, 2011 at 2:10 PM, satish patel satish...@hotmail.com wrote: Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S Shouldn't the phones re-register on their own? Mine do it every few minutes. -M -- _ -- 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] Restart asterisk destroy all registered SIP peers
Issue is we are running customer support queue and if by chance if i need to restart asterisk then they will not able to get call until phone get register :( Let me check polycom default timeout and set to min. -S From: mden...@gmail.com Date: Fri, 20 May 2011 15:03:35 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers On Fri, May 20, 2011 at 3:00 PM, satish patel satish...@hotmail.com wrote: We have polycom 501 and i am waiting since last 5 min no registration require appear. -S With Polycom 321 you can poke around the menus -- one of them has a countdown timer which will show you when the next registration happens. -M -- _ -- 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] Restart asterisk destroy all registered SIP peers
Hey Eric, I do have qualify=yes. Am i missing something ? [seb-exten](!) ; Template type=friend host=dynamic context=from-sip qualify=yes dtmfmode=rfc2833 nat=no cc_agent_policy=generic cc_monitor_policy=generic [7022](seb-exten) callerid=Rover Conference 7022 accountcode=Rover Conference mailbox=7022@default [7023](seb-exten) callerid=Faire Conference 7023 accountcode=Faire Conference mailbox=7023@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Fri, 20 May 2011 15:15:45 -0400 Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, May 20, 2011 3:10 PM To: asterisk-users Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers Issue is we are running customer support queue and if by chance if i need to restart asterisk then they will not able to get call until phone get register :( Let me check polycom default timeout and set to min. Asterisk should cache the registrations across a restart and reboot. I belive this feature was added in 1.4. You should not need to set a low registration timeout. If you set it because of NAT issues, setting qualify=yes will keep the translations open. -- _ -- 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] Restart asterisk destroy all registered SIP peers
There is a fix https://issues.asterisk.org/view.php?id=19318 -- Sent from my iPhone On May 20, 2011, at 4:40 PM, satish patel satish...@hotmail.com wrote: Hey Eric, I do have qualify=yes. Am i missing something ? [seb-exten](!) ; Template type=friend host=dynamic context=from-sip qualify=yes dtmfmode=rfc2833 nat=no cc_agent_policy=generic cc_monitor_policy=generic [7022](seb-exten) callerid=Rover Conference 7022 accountcode=Rover Conference mailbox=7022@default [7023](seb-exten) callerid=Faire Conference 7023 accountcode=Faire Conference mailbox=7023@default From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Fri, 20 May 2011 15:15:45 -0400 Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, May 20, 2011 3:10 PM To: asterisk-users Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers Issue is we are running customer support queue and if by chance if i need to restart asterisk then they will not able to get call until phone get register :( Let me check polycom default timeout and set to min. Asterisk should cache the registrations across a restart and reboot. I belive this feature was added in 1.4. You should not need to set a low registration timeout. If you set it because of NAT issues, setting qualify=yes will keep the translations open. -- _ -- 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 -- _ -- 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's zombie processes
Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- 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 -- _ -- 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] Static Vs Dynamic queue confusion
I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member = technology/ -- _ -- 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-cpu utilization 60 %
How much memory have allocate to VM ? and send top or ps command output. Date: Thu, 19 May 2011 22:44:58 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 Anyone else facing high CPU usage problem with Asterisk 1.6.2.13 or any Elastix 2.0.3 users here ? With just 3 concurrent calls and none in queue, the CPU is constantly above 40%. The moment CPU goes above 50%, calls start to break. I am a newbie and at lack of options... Sans -- _ -- 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] Static Vs Dynamic queue confusion
agents.conf agent = 7101,1234,Agent1 agent = 7102,1234,Agent2 queues.conf ... ... member = Agent/7201 member = Agent/7202 CLI output holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet Agent/7101 with penalty 1 (dynamic) (Unavailable) has taken no calls yet Agent/7102 with penalty 1 (dynamic) (Unavailable) has taken no calls yet No Callers agents are not getting calls. and what is Invalid ? From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 19 May 2011 16:41:02 + Subject: [asterisk-users] Static Vs Dynamic queue confusion I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member = technology/ -- _ -- 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] dahdi command not available
Thanks for reply Marcelo, I don't know what was the problem but after reboot machine it works! I am pretty sure i did service dahdi start/stop but that didn't work. -S Date: Thu, 19 May 2011 16:44:18 -0300 From: ellm...@freeddom.com To: isr...@gmail.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi command not available also, make sure that when you installed asterisk, the option to load the dahdi module was select. when you run a ./configure it scans your system and when you run make menuselect, the resource module dahdi will be marked to be compiled and installed :) --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: isr...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 May, 2011 3:48:05 PM Subject: Re: [asterisk-users] dahdi command not available Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- 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 -- _ -- 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] Agent (Invalid) has taken no calls yet
How to get rid on following.. why its Invalid ? holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -- _ -- 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] script to trim sip.conf
Hey Guys! Sorry i am posting scripting question in asterisk forum but i had no choice. also i am not script expert so i though anyone here might help me. following is my example sip.conf now i want to add accountcode=callerid_name for example accountcode=Katie Wilson in entire file. we have around 200 extension could someone help me to figure out how to do that with perl script or shell would be fine. [100](seb-exten) callerid=Katie Wilson 100 mailbox=100@default [200](seb-exten) callerid=Ramona Minero 200 mailbox=200@default -- _ -- 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] script to trim sip.conf
Holy cow! you made my day Thank you so much... It works great!!! S. From: mden...@gmail.com Date: Tue, 17 May 2011 17:02:55 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] script to trim sip.conf On Tue, May 17, 2011 at 4:21 PM, satish patel satish...@hotmail.com wrote: Hey Guys! Sorry i am posting scripting question in asterisk forum but i had no choice. also i am not script expert so i though anyone here might help me. following is my example sip.conf now i want to add accountcode=callerid_name for example accountcode=Katie Wilson in entire file. we have around 200 extension could someone help me to figure out how to do that with perl script or shell would be fine. [100](seb-exten) callerid=Katie Wilson 100 mailbox=100@default [200](seb-exten) callerid=Ramona Minero 200 mailbox=200@default Satish, Give this a shot: cat sip.conf | perl -pi -e s/^callerid=\(.*)\ (.*)/callerid=\\$1\ \$2\naccountcode=\\$1\/ sip.conf.new and compare them. -- _ -- 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] res_timing_timerfd.so Vs res_timing_dahdi.so
Thanks Leif, I had changed it to res_timing_dahdi and since last few days it seem good. -S Date: Sun, 15 May 2011 15:48:03 -0400 From: leif.mad...@asteriskdocs.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so On 11-05-13 11:39 AM, isr...@gmail.com wrote: I haven't tried with timerfd but with timer pthread 1.8 is very unstable I think I have seen a post to the list from kevin fleming that the same is for timerfd that there is a nasty bug which they haven't found the reason for yet My experience is that you should pretty much always use res_timing_dahdi unless you're on a platform on which you can't install DAHDI. You don't need any hardware to use timing from DAHDI because timing is generated by the kernel. My order of preference for stability is: * res_timing_dahdi * res_timing_timerfd * res_timing pthread The timerfd and pthread modules are relatively new, and sometimes people run into stability problems while using them. If you can use res_timing_dahdi I recommend you do so. Leif. -- _ -- 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-cpu utilization 60 %
Sorry fro hijacking thread. I have following process running on my asterisk eating around 2 or 3% CPU constantly. I knew events0/1 is CPU queue but why only single queue is busy ? I have kernel running preemtive with 1000Hz satish@campbx1:~$ ps aux | grep events root 9 1.7 0.0 0 0 ?SMay08 201:35 [events/0] root10 0.0 0.0 0 0 ?SMay08 1:19 [events/1] Date: Mon, 16 May 2011 17:37:16 +0300 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % On Mon, May 16, 2011 at 05:19:20PM +0430, Pezhman Lali wrote: check your running process, if you have more than one asterisk in your top re install your asterisk. Reinstall? Care to explain why? -- 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] Asterisk-cpu utilization 60 %
First grab LWP thread ID which is eating more CPU ps -LlFm -p `pidof asterisk` Now look into your asterisk.stack.txt and search particular LWP thread ID see following example Thread 10 (Thread 0x41d8f940 (LWP 3406)): #0 0x0033ce2ca436 in poll () from /lib64/libc.so.6 #1 0x004933c0 in ast_io_wait () #2 0x2aaabd9510cd in network_thread () #3 0x004f8b2c in dummy_start () #4 0x0033cee06367 in start_thread () from /lib64/libpthread.so.0 #5 0x0033ce2d2f7d in clone () from /lib64/libc.so.6 Now you have piece of cake. whatever the issue is you can find in above few lines.. -S Date: Mon, 16 May 2011 20:38:34 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Any finding in my asterisk.stack.txt ? 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 -- _ -- 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] dahdi command not available
Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI dahdi tab tab No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI root@campbx1:/etc/wanpipe# wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A102-SH : SLOT=2 : BUS=7 : IRQ=3 : CPU=A : PORT=1 : HWEC=64 : V=37 2 . AFT-A102-SH : SLOT=2 : BUS=7 : IRQ=3 : CPU=A : PORT=2 : HWEC=64 : V=37 Card Cnt: A101-2=1 root@campbx2:/etc/asterisk# lsmod Module Size Used by dahdi_echocan_mg2 5662 23 wanec 381336 0 af_wanpipe 34483 0 wanpipe 813623 1 wanrouter 52003 6 wanec,af_wanpipe,wanpipe sdladrv 221273 4 wanec,af_wanpipe,wanpipe,wanrouter dahdi 210313 2 dahdi_echocan_mg2,wanpipe crc_ccitt 1675 1 dahdi fbcon 39612 71 tileblit2487 1 fbcon font8053 1 fbcon bitblit 5875 1 fbcon softcursor 1565 1 bitblit -- _ -- 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-cpu utilization 60 %
Check this out http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ -- Sent from my iPhone On May 15, 2011, at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, May 15, 2011 at 08:24:08AM +0200, Leandro Dardini wrote: 2011/5/15 RSCL Mumbai rscl.mum...@gmail.com On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.com wrote: Check if someone is brute forcing your asterisk accounts. It used to happen to me before I install fail2ban. You can easily check the full log of asterisk or with just a tcpdump -i any -n port 5060 or port 4569. Thx for the tcpdump command. Checked, all looks good. Packets coming from trusted domains only. What should be the next step ? Thx Sans Have you tried to restart asterisk? As last chance, install strace and check what is asterisk doing. Get the pid (PID) of the running asterisk and run: strace -p PID -f -F /tmp/strace.log Not exactly. Asterisk is multi-threaded. strae traces a specific thread. To see the most active thread, press 'H' (shift-h) in top. Wait for the display to refresh at least twice (on the first time it won't make sense) and now check to see which is the top thread. -- 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] 1.8 and prematuremedia problem
Thanks and I did that and my figure are cross now. Let see -- Sent from my iPhone On May 15, 2011, at 8:35 AM, d tbsky tbs...@gmail.com wrote: hi: maybe you can try noload res_timing_timerfd in modules.conf and see what asterisk pick up for timing. in my system, if I disable res_timing_timerfd, then dahdi timing is selected and system become stable. Regards, tbskyd 2011/5/14 satish patel satish...@hotmail.com: You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch
Re: [asterisk-users] 1.8 and prematuremedia problem
Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php? id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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 -- _ -- 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
Re: [asterisk-users] 1.8 and prematuremedia problem
Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information
Re: [asterisk-users] 1.8 and prematuremedia problem
You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also
[asterisk-users] ConfBridge for 1.8 ?
Hey Guys! I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and install with 1.8 ? -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] ConfBridge for 1.8 ?
Holly Cow! Its there already sorry i thought it will only comes with 1.10. We are using meetme since last 5 year do you think confbridge is better then meetme ? just need your suggestion /usr/lib/asterisk/modules/app_confbridge.so From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 12 May 2011 14:33:12 + Subject: [asterisk-users] ConfBridge for 1.8 ? Hey Guys! I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and install with 1.8 ? -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 -- _ -- 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] ConfBridge for 1.8 ?
Thanks Kevin, Good to know. Different mean features vise or performance ? Do you think it is a good idea to replace meetme with confbridge in current 1.8 or i should wait for 1.10 ? -S Date: Thu, 12 May 2011 09:50:12 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ConfBridge for 1.8 ? On 05/12/2011 09:37 AM, satish patel wrote: Holly Cow! Its there already sorry i thought it will only comes with 1.10. We are using meetme since last 5 year do you think confbridge is better then meetme ? just need your suggestion /usr/lib/asterisk/modules/app_confbridge.so The app_confbridge in Asterisk 1.8 is very different from the one in trunk (what will become Asterisk 1.10). -- 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] Light indicator managed by Asterisk
Check out http://kb.smartvox.co.uk/index.php/asterisk/sip-extensions/shared-voicemail-part2/ Date: Thu, 12 May 2011 14:38:46 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Light indicator managed by Asterisk Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] how to reload agents.conf ?
How to reload only agents.conf ? -- _ -- 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 1.8 somehow dead
Guys! I am running 1.8 on production we have one PRI and 50 extensions. since last few days its working fine but today some how server load get high 194 % CPU and when i did asterisk -r i got CLI but no out put for any command. I check logs and nothing interesting there.. I am not using any advance feature just Voicemail, Meetme and calling.. Anybody having this kind of issue ? -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] 1.8 and prematuremedia problem
I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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 -- _ -- 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] 1.8 and prematuremedia problem
Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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 -- _ -- 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] iax2 Max retries exceeded to host
We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211) [May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3040332, seqno=212) [May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from 'sip:7...@laverne.east.ora.com' failed for '172.30.245.85:5060' - No matching peer found [May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from 'sip:7...@laverne.east.ora.com' failed for '172.30.245.85:5060' - No matching peer found [May 10 15:23:49] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 2, ts=3045385, seqno=213) [May 10 15:23:54] WARNING[2054]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3050332, seqno=214) [May 10 15:24:04] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3060332, seqno=215) [May 10 15:24:10] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 2, ts=3066385, seqno=216) [May 10 15:24:14] WARNING[2051]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3070332, seqno=217) -- _ -- 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] iax2 Max retries exceeded to host
campbx1*CLI iax2 show netstats
LOCAL -
REMOTE
Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost
% Drop OOO Kpkts FirstMsgLastMsg
IAX2/orasebcam-612 83 -10-1 -1 0 -1 00 40 0
0 00 0 Tx:NEW Tx:LAGRQ
IAX2/7504-1407204 -10-1 -1 0 -120200 0
0 00 0 Rx:NEW Tx:ACK
IAX2/orasebcam-3360 104 -10-1 -1 0 -1 50 40 0
0 00 0 Rx:NEW Rx:ACK
IAX2/orasebcam-828784 -10-1 -1 0 -12020 40 0
0 00 0 Tx:NEW Rx:ACK
IAX2/7504-15510 178 -10-1 -1 0 -1 200 0
0 00 0 Rx:NEW Tx:ACK
5 active IAX channels
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 10 May 2011 19:27:26 +
Subject: [asterisk-users] iax2 Max retries exceeded to host
We have IAX2 peer between two asterisk and I am getting following error
following IAX2 WARNING. IAX calling is functional
[May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11, ts=3030332, seqno=211)
[May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11, ts=3040332, seqno=212)
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register:
Registration from 'sip:7...@laverne.east.ora.com' failed for
'172.30.245.85:5060' - No matching peer found
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register:
Registration from 'sip:7...@laverne.east.ora.com' failed for
'172.30.245.85:5060' - No matching peer found
[May 10 15:23:49] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 2, ts=3045385, seqno=213)
[May 10 15:23:54] WARNING[2054]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11, ts=3050332, seqno=214)
[May 10 15:24:04] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11, ts=3060332, seqno=215)
[May 10 15:24:10] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 2, ts=3066385, seqno=216)
[May 10 15:24:14] WARNING[2051]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11, ts=3070332, seqno=217)
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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?
Thanks to all for reply, I have already put 1.8 in production. Actually we are using basic function so I hope we are good and fingurs cross. -- Sent from my iPhone On May 9, 2011, at 7:18 AM, Alec Davis siva...@paradise.net.nz wrote: Are you not seeing issues with *8 call pick up then ? -- Thanks, Phil https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues, particulary when you have pickupsounds enabled. Alec -- _ -- 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 1.8 latest branch safe for production ?
Which release are you running as this is still open https://issues.asterisk.org/view.php?id=18654 -- Thanks, Phil I am using current SVN branch 1.8 and We aren't using above call pickup features. _ -- 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] iax2 issue in asterisk
Hey guys! I have issue between iax vs iax2 following is my setup asterisk-1.2 --IAXAsterisk-1.8 I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8 -- _ -- 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
