Re: [asterisk-users] odbc error - server is gone
Isql is a process by which you can test your ODBC connection without calling to asterisk. Just one line command to test the ODBC connectivity. **echo select 1 | isql -v *asterisk-connector* here asterisk-connector is your OBDC connection name. On Sun, May 1, 2011 at 12:10 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: isql? On Sat, Apr 30, 2011 at 6:18 PM, Pezhman Lali l...@lopl.net wrote: check your odbc connection with isql best On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.comwrote: You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I believe 1.4.41 is current) and see if your issue has been resolved. Thanks, --Warren Selby, dCAP On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error (ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It * only ever really worked with MySQL. It will not work with any other * database, since most databases prepare their statements on the server, * and if you disconnect, you invalidate the statement handle. Hence, if * you disconnect, you're going to fail anyway, whether you try to execute * a second time or not. */ This function is used all over. On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com sherwood.mcgo...@gmail.com wrote: On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.com rizwanhas...@gmail.com wrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemailand herehttp://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage . I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users --
[asterisk-users] Retrieving sound files from DB as opposed to filesystem
Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if the performance of the Asterisk system might suffer in terms of latency etc. if we're to have it retrieve sound files from a database using odbc as opposed to storing them locally on the filesystem. Note, these are not prompts...these are sound files that are being created through a web-app and being stored in the DB as BLOB or similar datatype that's good/efficient to store audio/video files in a DB. We need these be made available through the asterisk system to play over the phone. Although the DB uses a SAN, the Asterisk System has no connectivity to the SAN but is connected on the same physical ethernet switch with a multi-Gbps backplane. The way the system is being designed, it's possible for us to end up with 000s of these sound files stored in the DB, not to mention several asterisk systems in a pool/cluster/farm requesting these files, so using the local filesystem might not be scalable or efficient. Any advice/comments/suggestions welcome :) -- _ -- 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 member invalid
Hi, I'm using asterisk version 1.8.3.3. In earlier versions I used queues, but with the new version the queuing mechanism doesn't work If I look in the CLI at I see that the queue-member is invalid: Members: DADHI/g3/0655871460 (Invalid) has taken no calls yet The queues.conf looks like this: [general] persistentmembers = yes monitor-type = MixMonitor [test] musicclass = default strategy = rrmemory member = DADHI/g3/0655871460 timeout = 60 retry = 1 maxlen = 5 If already changed the modules.conf to this, but with no success [modules] autoload=yes preload = pbx_config.so preload = pbx_ael.so preload = chan_local.so preload = app_queue.so noload = pbx_gtkconsole.so load = res_musiconhold.so noload = chan_alsa.so Does anybody have an idea what could be the problem? Best Regards, Arjan Kroon -- _ -- 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] HA Asterisk
On Mon, May 2, 2011 at 12:07 AM, Kaushal Shriyan kaushalshri...@gmail.comwrote: Hi Jim, Thanks for the explanation, I have couple of questions here. 1) Does the xorcom box support *8 Port PRI E1 Interface*. ? 2) Also the Primary and Secondary Asterisk Server can be any server which will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow) Application and customizable or do i also need to buy this from Xorcom ? Not sure i understand that. 3) How does the xorcom box communicate with the Asterisk Server which do not contain any PRI Card inside the system. Much Appreciated. Thanks and Regards, Kaushal Kaushal, 1) it's all clearly explained on their page. Looking at the video, one can tell they have 8 PRI ports on that box and 8 FXS ports and there's space for 3 further 8-channel modules that can be added. You can get an XR0111 for 8 PRIs (or XR0015 for BRI): http://www.xorcom.com/telephony-interfaces/astribank-models.html 2) It also states there that the Astribank's drivers have been a part of Zaptel/DAHDI since early 2006. Which means that it's MOST likely compatible with any home-baked Asterisk installation without the need to buy Xorcom Servers. 3) Lastly, it clearly uses these Astribank drivers in DAHDI to make the Astribank channel bank as an external hardware to Asterisk to talk back and forth. Since USB is a physical connection between the two, I'm sure if a server is down, the software in Astribank can detect the lack of connectivity on that USB port (i.e. voltage) as well as it might realise there's no communication between it and the Astribank driver in DAHDI on the Asterisk server. One should not just try and get answers the easy way. You could've figured all this out in 5 mins just like I did...not that I'm saying I'm really smart ;) Anyway, hope it helps :) Now, I wonder what're the alternatives that people have been using for Asterisk HA other than commercially available solutions like HAAST and Astribanks assuming that kaushal is right and SCF isn't production ready yet. Anyone wants to chime in here with a solution built with readily available linux software like heartbeat , linux-ha, shared filesystems, filesystem replication and of course asterisk realtime? My requirement might be more along the lines of having several asterisk servers in a farm/pool without actually caring about the failover, so it might not even matter for me to worry about all of this, but I'm still curious as to what people are doing out there. Cheers -- _ -- 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] default context overrides context of peer
Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to 'defcontext' and the calls fail. After the peer registers 'sip show peer peername' show the context as 'outcontext', but while making a call the default context in sip.conf overrides the peer context. Is there any other setting that I need to do in asterisk 1.6.2.17? Thanks -- _ -- 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] out of the blue one way audio
Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel(R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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] out of the blue one way audio
Check if this problem happening with xlite useres only i remember there is option in xlite causing this problem On May 2, 2011 2:36 PM, Tarek Sawah tareksa...@hotmail.com wrote: -- _ -- 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] Password to be ecrypted?
On Tue, Apr 26, 2011 at 04:03:51PM +0100, A J Stiles wrote:
On Tuesday 26 Apr 2011, bilal ghayyad wrote:
Hi All;
I am using Asterisk 1.8, how I can protect my self from hackers in case
they was able to see my sip.conf file? I need the password to be encrypted,
how?
Short answer: You can't. Asterisk itself needs to be able to read the
stored
passwords. The Source Code to Asterisk is readily available. Therefore,
anyone who can read sip.conf, even if it is encrypted, will necessarily be
able to decrypt it.
Slightly more helpful answer: Make sure that sip.conf can only be read by
the
root user;
# chown root:root /etc/asterisk/sip.conf
# chmod 600 /etc/asterisk/sip.conf
This is about as safe as it gets. If somebody manages to get root access to
your Asterisk box, then you're already shafted .
This implies running Asterisk as root, which is certainly not the safest
thing to do.
chown asterisk /etc/asterisk/sip.conf
chmod 600 /etc/asterisk/sip.conf
If you really want to split out the secret part, you can have something
along the lines of:
sip.conf:
[general]
;host, port, and such
[phone1]
; Everything, besides 'secret'
[trunk1]
; Everything, besides 'secret'
#include sip_secret.conf
sip_secret.conf:
[general](+)
register = ...
[phone1](+)
secret = ...
[trunk1](+)
secret = ...
This way only sip_secret.conf needs to be kept confidential.
But then again, anyone with access to asterisk should be able to read
the configuration ('sip show users', GetConfig in the manager interface,
whatever).
There are further obfuscations to be done (there has been a previous
thread about this subject). But you should first clarify (to yourself,
mostly) what is the threat you want to protect your system from. Given
enough resources, the NSA will get those passwords anyway
(http://xkcd.com/538/ ). But you should make good security to protect
your system from reasonable threats.
--
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
--
_
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Re: [asterisk-users] out of the blue one way audio
this is happening on all Soft phones are facing the same problem. Zoiper , X=lite , our own pjsip based dialer (CRM). this was not the issue .. it happened suddenly .. we switched internet links even. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 2 May 2011 14:45:58 +0300 From: hatemm...@gmail.com To: asterisk-users@lists.digium.com CC: yamennaj...@ids-tech.net Subject: Re: [asterisk-users] out of the blue one way audio Check if this problem happening with xlite useres only i remember there is option in xlite causing this problem On May 2, 2011 2:36 PM, Tarek Sawah wrote: -- _ -- 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] out of the blue one way audio
because they are behind a router and using private IP addresses. and the Cisco router is Nating our traffic Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 08:11:23 -0400 Subject: Re: [asterisk-users] out of the blue one way audio Why nat=yes ? -- Sent from my iPhone On May 2, 2011, at 7:33 AM, Tarek Sawah wrote: Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel (R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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] Asterisk repository: asterisk14-addons-mysql
Hello, We have chosen to upgrade our Trixbox installations (2.6.2.3, asterisk 1.4.20) and everything work smooth. The problem we face now is that asterisk14-addons-mysql looks to have not been compiled with uniqueID feature and we are asking your opinion about what should be the best fix for this problem. Our workarround was to overwrite (from backup) the cdr_addon_mysql.so module, but this is not the best approach as in case for future updates (yes - we know support of 1.4 have reach it's end but who knows). On the other hand we could try to compile from sources but this procedure we have tried to avoid when we choose to use asterisk.org repository. My idea is to ask for an additional addons-mysql RPM package - with uniqueID enabled - but I do not know exactly where I have to post this question. What is your opinion? Best regards, Ioan. -- _ -- 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] HA Asterisk
On Fri, Apr 29, 2011 at 7:29 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) The current production ready versions of Asterisk (1.4, 1.6, 1.8) do not have any native HA support. You have to engineer that on your own, or purchase a commercial product that handles it for you. How this is engineered would be based on your specific requirements. Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? AsteriskNOW is a simple to install complete Asterisk setup, just add hardware. While that is great, it would probably be more of a pain to make AsteriskNOW into an HA install than build one yourself based on your specific requirements. I haven't personally tried though, so YMMV. It appears that AsteriskNOW 1.7.1 64-bit contains Asterisk version 1.4.35 and 1.6.2.11. Both versions are now at Security Update Only status (but that's a conversation for another thread) We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. In my specific experience, I would say Asterisk is neither CPU or Memory intensive. Memory has never been an issue, and we are not transcoding between different codecs. If you plan to do a lot of transcoding in software, then your CPU usage will increase. You would have to test using your specific requirements to know how it will impact your systems. -Jonathan -- _ -- 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] HA Asterisk
On Mon, May 2, 2011 at 1:10 AM, A E [Gmail] all.efor...@gmail.com wrote: Now, I wonder what're the alternatives that people have been using for Asterisk HA other than commercially available solutions like HAAST and Astribanks assuming that kaushal is right and SCF isn't production ready yet. Anyone wants to chime in here with a solution built with readily available linux software like heartbeat , linux-ha, shared filesystems, filesystem replication and of course asterisk realtime? My requirement might be more along the lines of having several asterisk servers in a farm/pool without actually caring about the failover, so it might not even matter for me to worry about all of this, but I'm still curious as to what people are doing out there. For our specific needs we have build an active/passive Asterisk cluster based on CentOS 5 and cman/drbd/gfs2. Two nodes replicate data (configs, voicemail, provisioning data) on a Master/Master DRBD volume, using GFS2 as the shared file system. We use Asterisk Realtime via ODBC (MySQL Backend) for SIP/Extensions/CDR. All services bind to a floating IP Address. CMAN controls what server is running the services at any time, and handles migrating of the IP as well. Lights Out cards (via IPMI) are used for fencing. For access to the PSTN, I prefer to use an external device. We run a mix of Cisco 2800's and AudioCodes Mediant 1000's. I prefer to use PSTN to SIP gateways over cards built-in to the servers, or Astribanks as I feel they are more flexible. You could allow direct media, or allow multiple servers to communicate with the gateways at that same time. So that is the setup that we have chosen, and it might not be right for anyone else. The best advice I can give is to implement something at your comfort level, and test test test! I am aware of the potential issues with our setup, and am prepared to deal with them because of extensive testing. -Jonathan -- _ -- 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] HA Asterisk
On Sun, May 1, 2011 at 3:03 AM, Terry Brummell te...@brummell.net wrote: 8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No messing around with switches and cables an crap. I agree, use a SIP Gateway. The AudioCodes Mediant 1000 supports up to 4 T1/E1/J1, so use two of them. That also keeps you going in case one of the gateways dies. -Jonathan -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
On Wed, 2011-04-27 at 21:34 +0200, Olle E. Johansson wrote: Friends, We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security patches would be committed. This is already a delay from the original plan published by Russell Bryant. Unfortunately, I think this is way too early. My feeling and experience is that 1.8 is not ready for production in the environments I work in - large scale installations. Customers are not planning migration and all new installs are still 1.4. Tests we've been doing with 1.8 has failed within just a short time and so badly that customers has not paid me to spend any further time with 1.8. Just a thought If Digium / the community realy want an objective way of deciding whether can/should migrate to any other version, you realy need a feature-matrix (pethaps starting from version 1.2.*) And for every and each version a statement if it is: - discontinued - tested - test finalized, result indicating it is fully and identically functional - test finalized, result indicating that this feature is changed in either behaviour of configuration - not yet tested. I realize it is quite a job to do, but if done it would be for everyone easily to see if it is worthwhile to start migrating. Anyway for both documentation purposes and bugtracking it would be nice if each and every feature has a unique numerique identifier. And perhaps there is a fair chance that the people from the quality department at Digium already have such a list. 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
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote: Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? You can intruduce a predefined amount of distortion on your ip-connection (packet loss, fluctuating delay, out of secuence reception of packets, limited bandwith) All of these will have a serious impact on your VOIP-connection. See lartc about it. Good thing about it, is that you pre-define how bad a line is, and it produces re-producable results 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
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
2 maj 2011 kl. 18.09 skrev Hans Witvliet: On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote: Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? You can intruduce a predefined amount of distortion on your ip-connection (packet loss, fluctuating delay, out of secuence reception of packets, limited bandwith) All of these will have a serious impact on your VOIP-connection. See lartc about it. Good thing about it, is that you pre-define how bad a line is, and it produces re-producable results I use a laptop with a usb-ethernet connected in bridge mode as a voip destroyer. Using TC you can inject a lot of bad stuff on the connection. /O -- _ -- 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] music on hold skipping
For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -- _ -- 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 call completion issue
Hi All, I am testing CC feature with asterisk 1.8 but i am having some issue. We have polycom 501 SIP phone and those are configured with two line with same extensions. When i am requesting for CC i am not getting call back from asterisk but it works if i reboot my polycom phone ( In short when phone get register ) Is this because of two line configured ? or some configuration issue ? -- _ -- 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 completion issue
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 12:19 PM To: asterisk-users Subject: [asterisk-users] asterisk call completion issue Hi All, I am testing CC feature with asterisk 1.8 but i am having some issue. We have polycom 501 SIP phone and those are configured with two line with same extensions. When i am requesting for CC i am not getting call back from asterisk but it works if i reboot my polycom phone ( In short when phone get register ) Is this because of two line configured ? or some configuration issue ? [Danny Nicholas] I would check call-limit and see what reducing that would do for you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk call completion issue
I have call-limit=1 at sip.conf From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 12:20:40 -0500 Subject: Re: [asterisk-users] asterisk call completion issue From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 12:19 PM To: asterisk-users Subject: [asterisk-users] asterisk call completion issue Hi All, I am testing CC feature with asterisk 1.8 but i am having some issue. We have polycom 501 SIP phone and those are configured with two line with same extensions. When i am requesting for CC i am not getting call back from asterisk but it works if i reboot my polycom phone ( In short when phone get register ) Is this because of two line configured ? or some configuration issue ? [Danny Nicholas] I would check call-limit and see what reducing that would do for 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
Re: [asterisk-users] [SOLVED] asterisk call completion issue
After adding callcounter=yes at sip.conf it works! Cheers! From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:34:32 + Subject: Re: [asterisk-users] asterisk call completion issue I have call-limit=1 at sip.conf From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 12:20:40 -0500 Subject: Re: [asterisk-users] asterisk call completion issue From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 12:19 PM To: asterisk-users Subject: [asterisk-users] asterisk call completion issue Hi All, I am testing CC feature with asterisk 1.8 but i am having some issue. We have polycom 501 SIP phone and those are configured with two line with same extensions. When i am requesting for CC i am not getting call back from asterisk but it works if i reboot my polycom phone ( In short when phone get register ) Is this because of two line configured ? or some configuration issue ? [Danny Nicholas] I would check call-limit and see what reducing that would do for 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 -- _ -- 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 call completion issue
If I recall correctly, callcounter supercedes call-limit in 1.8. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 12:57 PM To: asterisk-users Subject: Re: [asterisk-users] [SOLVED] asterisk call completion issue After adding callcounter=yes at sip.conf it works! Cheers! _ From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:34:32 + Subject: Re: [asterisk-users] asterisk call completion issue I have call-limit=1 at sip.conf _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 12:20:40 -0500 Subject: Re: [asterisk-users] asterisk call completion issue _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 12:19 PM To: asterisk-users Subject: [asterisk-users] asterisk call completion issue Hi All, I am testing CC feature with asterisk 1.8 but i am having some issue. We have polycom 501 SIP phone and those are configured with two line with same extensions. When i am requesting for CC i am not getting call back from asterisk but it works if i reboot my polycom phone ( In short when phone get register ) Is this because of two line configured ? or some configuration issue ? [Danny Nicholas] I would check call-limit and see what reducing that would do for 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 -- _ -- 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] Retrieving/Streaming audio/video files from DB using over AGI
Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( -- _ -- 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] Retrieving/Streaming audio/video files from DBusing over AGI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 02, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( [Danny Nicholas] In your original scenario you were opening yourself to probable latency issues - I would personally pursue something along the line of option B where I put the DB data into a temp file and ran a daemon to clear the temp files hourly or daily as needed. If the delivery worked well across most LAN's/WAN's, some gung-ho developer would have hosed another part of Asterisk trying to get that bell and whistle into the trunk. -- _ -- 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] Retrieving/Streaming audio/video files from DBusing over AGI
On Mon, May 2, 2011 at 2:30 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail] *Sent:* Monday, May 02, 2011 1:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( *[Danny Nicholas] * *In your original scenario you were opening yourself to probable latency issues – I would personally pursue something along the line of option B where I put the DB data into a temp file and ran a daemon to clear the temp files hourly or daily as needed. If the delivery worked well across most LAN’s/WAN’s, some gung-ho developer would have hosed another part of Asterisk trying to get that “bell and whistle” into the trunk.* Thanks Danny. I'm not so sure, that latency will be that much of an issue being on the same physical GbE switch as the DB server without any other traffic on it but sure, I know that a long time ago when I implemented Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess we do need to use that AGI AddOn of PUT SOUNDFILE after all. Would be good if more people can throw a few ideas around to see if there's a smarter way to do it. Another idea we had was to dumb these files (since they'll be very small in duration and thus in size) into a directory, run a web-server and have AGI retrieve them using curl and just use Background to play it. Thoughts? -- _ -- 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] Retrieving/Streaming audio/video files from DBusing over AGI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 02, 2011 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI On Mon, May 2, 2011 at 2:30 PM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 02, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( [Danny Nicholas] In your original scenario you were opening yourself to probable latency issues - I would personally pursue something along the line of option B where I put the DB data into a temp file and ran a daemon to clear the temp files hourly or daily as needed. If the delivery worked well across most LAN's/WAN's, some gung-ho developer would have hosed another part of Asterisk trying to get that bell and whistle into the trunk. Thanks Danny. I'm not so sure, that latency will be that much of an issue being on the same physical GbE switch as the DB server without any other traffic on it but sure, I know that a long time ago when I implemented Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess we do need to use that AGI AddOn of PUT SOUNDFILE after all. Would be good if more people can throw a few ideas around to see if there's a smarter way to do it. Another idea we had was to dumb these files (since they'll be very small in duration and thus in size) into a directory, run a web-server and have AGI retrieve them using curl and just use Background to play it. Thoughts? [Danny Nicholas] IMO, adding curl to the mix is just going to introduce another possible point of failure. If they are that small, why not do a daemonized delivery system? -- _ -- 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] Retrieving/Streaming audio/video files from DBusing over AGI
On Mon, May 2, 2011 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail] *Sent:* Monday, May 02, 2011 1:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( *[Danny Nicholas] * *In your original scenario you were opening yourself to probable latency issues – I would personally pursue something along the line of option B where I put the DB data into a temp file and ran a daemon to clear the temp files hourly or daily as needed. If the delivery worked well across most LAN’s/WAN’s, some gung-ho developer would have hosed another part of Asterisk trying to get that “bell and whistle” into the trunk.* Thanks Danny. I'm not so sure, that latency will be that much of an issue being on the same physical GbE switch as the DB server without any other traffic on it but sure, I know that a long time ago when I implemented Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess we do need to use that AGI AddOn of PUT SOUNDFILE after all. Would be good if more people can throw a few ideas around to see if there's a smarter way to do it. Another idea we had was to dumb these files (since they'll be very small in duration and thus in size) into a directory, run a web-server and have AGI retrieve them using curl and just use Background to play it. Thoughts? *[Danny Nicholas] * *IMO, adding curl to the mix is just going to introduce another possible point of failure. If they are that small, why not do a daemonized delivery system?* By daemonized delivery system, I'm assuming you mean have some background process running to transport these files from the DB to the asterisk server and play them? There are two issues with that a) Sounds like too much I/O esp. with small files getting written and deleted. b) What if there are several asterisk servers and the call can come into any of the servers. Do we invoke the daemon at will, run a SQL query, extract it from the DB, and transfer it to the asterisk server which initiated the request and then play it? Sounds like it might add a bit more latency than streaming it right inside the connection opened by AGI itself, although we could not store these files in the DB and just have them sit on a dedicated SAN volume and whenver a request comes in, we send it to the requesting asterisk server. That's all of course if I understood you correctly. -- _ -- 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] Retrieving/Streaming audio/video files from DBusing over AGI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 02, 2011 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI On Mon, May 2, 2011 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 02, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI Just realised that this can better be described another way: What we're essentially trying to do is be able to do any one of these a) stream an audio/video file stored in the DB via AGI into the current channel so that it plays on the phone OR b) Do something like what Realtime Voicemail does, where it gets the file from the DB, saves as a temp file in the user mailbox directory and then plays it to the caller but this needs to happen through AGI, something along the lines of readsql (a la func_odbc) inside of AGI OR c) Anything else that's better than a) and b) above that someone can suggest. P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which seems to be the only solution we can think of right now, other than of course having the DB machine exporting the SAN volume as an NFS share for the Asterisk server to mount, but that sounds like it'll be bad for performance? Thanks again No takers? :( [Danny Nicholas] In your original scenario you were opening yourself to probable latency issues - I would personally pursue something along the line of option B where I put the DB data into a temp file and ran a daemon to clear the temp files hourly or daily as needed. If the delivery worked well across most LAN's/WAN's, some gung-ho developer would have hosed another part of Asterisk trying to get that bell and whistle into the trunk. Thanks Danny. I'm not so sure, that latency will be that much of an issue being on the same physical GbE switch as the DB server without any other traffic on it but sure, I know that a long time ago when I implemented Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess we do need to use that AGI AddOn of PUT SOUNDFILE after all. Would be good if more people can throw a few ideas around to see if there's a smarter way to do it. Another idea we had was to dumb these files (since they'll be very small in duration and thus in size) into a directory, run a web-server and have AGI retrieve them using curl and just use Background to play it. Thoughts? [Danny Nicholas] IMO, adding curl to the mix is just going to introduce another possible point of failure. If they are that small, why not do a daemonized delivery system? By daemonized delivery system, I'm assuming you mean have some background process running to transport these files from the DB to the asterisk server and play them? There are two issues with that a) Sounds like too much I/O esp. with small files getting written and deleted. b) What if there are several asterisk servers and the call can come into any of the servers. Do we invoke the daemon at will, run a SQL query, extract it from the DB, and transfer it to the asterisk server which initiated the request and then play it? Sounds like it might add a bit more latency than streaming it right inside the connection opened by AGI itself, although we could not store these files in the DB and just have them sit on a dedicated SAN volume and whenver a request comes in, we send it to the requesting asterisk server. That's all of course if I understood you correctly. [Danny Nicholas] For my .02, I would run the daemon on the central server and just push out changed files - should be insignificant latency/overhead. If you wanted to run daemons, on every client, you could do that, but why? -- _ -- 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] sip busy detect
Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid=Guest 7527 mailbox=7527@default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:2] Goto(SIP/7604-0006, s-CONGESTION,1) in new stack -- Goto (macro-stdexten,s-CONGESTION,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] sip busy detect
Remove your call-limit or increase your calllimit above your busy level -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 4:56 PM To: asterisk-users Subject: [asterisk-users] sip busy detect Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid=Guest 7527 mailbox=7527@default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:2] Goto(SIP/7604-0006, s-CONGESTION,1) in new stack -- Goto (macro-stdexten,s-CONGESTION,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] sip busy detect
Thanks for reply, I had tried to increase call-limit=2 or more also removed and in that case i am hearing ringing not detecting busy channel :( From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 16:59:10 -0400 Subject: Re: [asterisk-users] sip busy detect Remove your call-limit or increase your calllimit above your busy level -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 4:56 PM To: asterisk-users Subject: [asterisk-users] sip busy detect Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid=Guest 7527 mailbox=7527@default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:2] Goto(SIP/7604-0006, s-CONGESTION,1) in new stack -- Goto (macro-stdexten,s-CONGESTION,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 -- _ -- 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] sip busy detect
We always rely on our phones to send back a busy when busy. Is there a reason you can't do that? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 5:04 PM To: asterisk-users Subject: Re: [asterisk-users] sip busy detect Thanks for reply, I had tried to increase call-limit=2 or more also removed and in that case i am hearing ringing not detecting busy channel :( From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 16:59:10 -0400 Subject: Re: [asterisk-users] sip busy detect Remove your call-limit or increase your calllimit above your busy level -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 4:56 PM To: asterisk-users Subject: [asterisk-users] sip busy detect Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid=Guest 7527 mailbox=7527@default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:2] Goto(SIP/7604-0006, s-CONGESTION,1) in new stack -- Goto (macro-stdexten,s-CONGESTION,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 -- _ -- 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] sip busy detect
We have polycom 501 phone. Do you know how to configure it to send back busy signal ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:07:22 -0400 Subject: Re: [asterisk-users] sip busy detect We always rely on our phones to send back a busy when busy. Is there a reason you can't do that? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 5:04 PM To: asterisk-users Subject: Re: [asterisk-users] sip busy detect Thanks for reply, I had tried to increase call-limit=2 or more also removed and in that case i am hearing ringing not detecting busy channel :( From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 16:59:10 -0400 Subject: Re: [asterisk-users] sip busy detect Remove your call-limit or increase your calllimit above your busy level -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 4:56 PM To: asterisk-users Subject: [asterisk-users] sip busy detect Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid=Guest 7527 mailbox=7527@default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:2] Goto(SIP/7604-0006, s-CONGESTION,1) in new stack -- Goto (macro-stdexten,s-CONGESTION,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 -- _ -- 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] sip busy detect
We use the following in the Polycom config files. call call.callsPerLineKey=1 / This will allow one call per line key on the phone, when calls are on all the line keys, the phone will return a busy. This will vary slightly if you use a different registration for each line key, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 5:14 PM To: asterisk-users Subject: Re: [asterisk-users] sip busy detect We have polycom 501 phone. Do you know how to configure it to send back busy signal ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:07:22 -0400 Subject: Re: [asterisk-users] sip busy detect We always rely on our phones to send back a busy when busy. Is there a reason you can't do that? -- _ -- 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] sip busy detect
Great! let me try.. We have same extension configured on two line. is this option will allow call transfer and two way conference ? See this thread http://forums.digium.com/viewtopic.php?t=3716 -S From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:18:02 -0400 Subject: Re: [asterisk-users] sip busy detect We use the following in the Polycom config files. call call.callsPerLineKey=1 / This will allow one call per line key on the phone, when calls are on all the line keys, the phone will return a busy. This will vary slightly if you use a different registration for each line key, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 5:14 PM To: asterisk-users Subject: Re: [asterisk-users] sip busy detect We have polycom 501 phone. Do you know how to configure it to send back busy signal ? From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 17:07:22 -0400 Subject: Re: [asterisk-users] sip busy detect We always rely on our phones to send back a busy when busy. Is there a reason you can't do that? -- _ -- 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] ATA refuses to answer a call?
I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the internet without any firewalling. - When people call this extension, the console shows that Asterisk accepts the call from the DAHDI channel, executes the SIP call, then... nothing. It either waits until the timeout set in the dialplan is up, then goes to voicemail (next step), or it sends a 'hangup cause 102' to the DAHDI channel. Conspicuously missing is the console saying SIP/username is ringing. The following is redacted output from such a call: -- Executing [6045551212@local:1] Dial(DAHDI/6-1, SIP/sipuser|20) in new stack -- Called sipuser -- Accepting call from '7785550001' to '6045551212' on channel 0/6, span 1 -- Channel 0/6, span 1 got hangup, cause 102 == Spawn extension (local, 6045551212, 1) exited non-zero on 'DAHDI/6-1' -- Hungup 'DAHDI/6-1' -- No one is available to answer at this time (1:0/0/0) -- _ -- 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] Retrieving sound files from DB as opposed to filesystem
Just from my experience with different DBs, stay away from BLOB data types as much as possible. On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if the performance of the Asterisk system might suffer in terms of latency etc. if we're to have it retrieve sound files from a database using odbc as opposed to storing them locally on the filesystem. Note, these are not prompts...these are sound files that are being created through a web-app and being stored in the DB as BLOB or similar datatype that's good/efficient to store audio/video files in a DB. We need these be made available through the asterisk system to play over the phone. Although the DB uses a SAN, the Asterisk System has no connectivity to the SAN but is connected on the same physical ethernet switch with a multi-Gbps backplane. The way the system is being designed, it's possible for us to end up with 000s of these sound files stored in the DB, not to mention several asterisk systems in a pool/cluster/farm requesting these files, so using the local filesystem might not be scalable or efficient. Any advice/comments/suggestions welcome :) -- _ -- 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] best current version and motherboard/CPU compatibilities
I've been away from asterisk for a while since 1.4.16 and only installed 1.6 once to run a test... can someone recommend what the best version to install is and the recommended CPU/motherboard for an * box these days? I'm just running about 20 handsets and 4-8 lines with POTS SIP mix. I remember there were some issues with bios a while back and a TDM card was required for timing conferencing, etc... are these requirements still an issue? I want to setup another * box and was wondering which CPU/motherboard to select... thanks, daveC -- SJREIA South Jersey Real Estate Investors Association Want to invest in Real Estate? come out and join over 450 real estate investors http://www.SJREIA.org Licensed NJ Real Estate Agent Buy This House REALTORs david.cant...@ibsonecall.com Mobile (856)813-7098 Office (856)324-4488 Pers Fax (646)827-7108 Ofc Fax (888)487-7711 Interlocking Business Solutions, LLC david.cant...@ibsonecall.com (856)581-8971 Home of the Videophone2009.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem
On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote: Just from my experience with different DBs, stay away from BLOB data types as much as possible. Hi CF, any particular reason why? I've had a good experience with it, in fact that's recommended by DB developers when it's a case of small files. They say only larger files greater than 500K-1MB should be stored on the filesystem using filestream or similar etc. Although at this point, this might be a moot point, as so far no one's been able to suggest a way to be able to stream the content of the BLOB field to Asterisk over the AGI connection into the current channel, such that Asterisk can just play it on the fly. We'll have to just go with getting the file to the requesting * server and then play it -- _ -- 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 debug MixMonitor misbehaviour
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually shows
the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for
things like privilege issues or filename issues?
**I had this working at one point and then stopped working. Not sure what I
changed.
System Info:
Asterisk 1.4.21.2
Queuemetrics 1.6.3.0
[queuedial]
; this piece of dialplan is just a calling hook into the [qm-queuedial]
context that actually does the
; outbound dialing - replace as needed - just fill in the same variables.
exten = _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
exten = _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
exten = _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
exten = _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
exten = _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
*exten = _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
exten = _XXX.,n,Goto(qm-queuedial,s,1)
CLI output:
-- Called 4904166356574@queuedial/n
-- Executing [4904166356574@queuedial:1]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_QUEUE=q-490) in new
stack
-- Executing [4904166356574@queuedial:2]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_NUMBER=4166356574) in
new stack
-- Executing [4904166356574@queuedial:3]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_AGENT=Agent/19053640558) in new stack
-- Executing [4904166356574@queuedial:4]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_CHANNEL=ZAP/g0/4166356574) in new stack
-- Executing [4904166356574@queuedial:5]
Set(Local/4904166356574@queuedial-d851,2, QueueName=q-490) in new stack
*-- Executing [4904166356574@queuedial:6]
MixMonitor(Local/4904166356574@queuedial-d851,2,
Q-q-490-1304399098.18.WAV|b|) in new stack*
-- Executing [4904166356574@queuedial:7]
Goto(Local/4904166356574@queuedial-d851,2, qm-queuedial|s|1) in new
stack
-- Goto (qm-queuedial,s,1)
Trying to locate file:
root@pbx:~ $ updatedb
root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
I also turned on the Debug but I couldn't see anything out of the norm. As
you can see above the CLI output is just fine.
Thanks,
Bruce
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Re: [asterisk-users] Asterisk repository: asterisk14-addons-mysql
[Danny Nicholas] IMO, one of the selling points of the add-on modules is that they can be compiled/tweaked without too much input from the base installation. I don't think you're going to get too far with the new/modified RPM request. Well - it looks we are the only ones needing that RPM with uniqueID enabled. IMO it is a pity that this feature could not be activated through configuration files (like cdr_mysql.conf for example). Thanks, Ioan -- _ -- 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] ATA refuses to answer a call?
On Tue, May 3, 2011 at 2:50 AM, Ernie Dunbar maill...@lightspeed.ca wrote: I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the internet without any firewalling. - When people call this extension, the console shows that Asterisk accepts the call from the DAHDI channel, executes the SIP call, then... nothing. It either waits until the timeout set in the dialplan is up, then goes to voicemail (next step), or it sends a 'hangup cause 102' to the DAHDI channel. Conspicuously missing is the console saying SIP/username is ringing. The following is redacted output from such a call: -- Executing [6045551212@local:1] Dial(DAHDI/6-1, SIP/sipuser|20) in new stack -- Called sipuser -- Accepting call from '7785550001' to '6045551212' on channel 0/6, span 1 -- Channel 0/6, span 1 got hangup, cause 102 == Spawn extension (local, 6045551212, 1) exited non-zero on 'DAHDI/6-1' -- Hungup 'DAHDI/6-1' -- No one is available to answer at this time (1:0/0/0) -- _ -- 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 Do: sip set debug ip IP OF DEVICE and see what it sends to 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
Re: [asterisk-users] default context overrides context of peer
On Mon, May 2, 2011 at 1:09 PM, Deepesh D deep.d2...@gmail.com wrote: Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to 'defcontext' and the calls fail. After the peer registers 'sip show peer peername' show the context as 'outcontext', but while making a call the default context in sip.conf overrides the peer context. Is there any other setting that I need to do in asterisk 1.6.2.17? Thanks -- _ -- 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 I am having the same issue and have not found a fix. Maybe we are both doing something wrong ;) -- _ -- 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
