[asterisk-users] Parked calls drop asterisk-1.4.22.1
Hi We are facing a problem for orphaned parked calls, we have the following config: asterisk -1.4.22.1 dahdi-linux-complete-2.2.0.2+2.2.0 and when we get an incoming call and after it gets parked, after some set time (here its 2 min), it goes back to the operator, but the problem is that randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the extension number of the operator) and we get the error as Unable to create the channel of type SIP (cause code 20) and then the call drops, we even tried asterisk-1.4.23.2, but in that version we were having problems with paging/intercom using the phones. [Oct 19 11:55:28] VERBOSE[2996] logger.c: == Timeout for SIP/5060-b781fe80 parked on 71. Returning to park-dial,SIP/5060,1 [Oct 19 11:55:28] VERBOSE[14641] logger.c: -- Executing [SIP/5...@park-dial:1] Dial(SIP/5060-b781fe80, SIP/5060|30|t) in new stack [Oct 19 11:55:28] WARNING[14641] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Oct 19 11:55:28] VERBOSE[14641] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Oct 19 11:55:28] VERBOSE[14641] logger.c: == Auto fallthrough, channel 'SIP/5060-b781fe80' status is 'CHANUNAVAIL' We also have the option of Page/Intercom through the phones that auto answer. Can any one share any ideas or opinions? Thank you, Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Calls stuck in the queue even when ext's are available
Hi.. We are facing a problem that is making the channel to be stuck. we are using asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues and one has 2 agents and the other 5 agents, from last week the second queue's channel is getting stuck, it happened 3 times till now and the problem is calls come into the queue and just the calls will be in the queue and will not ring any agents (static) even when are available..so when i went to the CLI and saw few channels were stuck: SIP/5060-b65171708...@ext-queues:11 Up Queue(8002|t||) SIP/5060-b65110708...@ext-queues:11 Up Queue(8002|t||) SIP/5060-b6515be08...@ext-queues:11 Up Queue(8002|t||) SIP/5060-0854ba808...@ext-queues:11 Up Queue(8002|t||) SIP/5060-08584ad08...@ext-queues:11 Up Queue(8002|t||) Even when i did soft hangup it did not hangupso i had to kill the asterisk process and had to restart it..i was researching and found that there is autofill=yes option that i am going to try it.please share if you have any thoughts in regards to the queue problem... Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Tones of dtmf during call
Hi, We are having the redial dtmf tones issue generated randomly in the Voip/SIP calls, [versions: asterisk 1.4.21.2, dahdi 2.0.2.2] and we have dtmf as inband in the trunks. We actually have mutiple locations one server at our datacenter, and from those locations people are complaining that they are having this random dtmf redialed kind of tones randomly may be like once/twice a day.Also I could see in the DTMF enabled logs that there was no generation of these DTMF digits...can anyone share some thoughts in regards to this, I appreciate your help..is there any thing that may be generated by the routers or is it the asterisk that may be causing this? Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Redial dtmf tones randomly...asterisk 1.4.21.2
We dont have any Digium cards, we just have a GrandStream FXS 8-port device with 2 analog phones and one Grand stream FXO 8-port device with one POTS line and both are connected to the netgear switchvery rarely the analog phones are used and its very rare that calls are made through POTS using FXO.we get this DTMF problem with the SIP phones(when called out or when we receive a call randomly)..I will try to capture the dtmf from the asterisk console with higher verbosity mode and also set the relaxeddtmf parameter. Thanks Sandesh On Thu, Jul 22, 2010 at 3:48 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: I would appreciate it if you didn't top-post. das sandesh sandesh...@gmail.com writes: Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... How do you interface with the PSTN? A Digium card? Either way you may want relaxdtmf=no in dahdi.conf if you don't have that already. You can see the DTMF happening on the Asterisk console if you set verbosity high enough. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Redial dtmf tones randomly...asterisk 1.4.21.2
Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the pathAlso we have forced the dtmf of the fxs port to be rfc2833. In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Thank you Sandesh attachment: wireshark.bmp-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Redial dtmf tones randomly...asterisk 1.4.21.2
Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... Thank you Sandesh On Wed, Jul 21, 2010 at 2:31 PM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: das sandesh sandesh...@gmail.com writes: In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Which end hears the DTMF, the SIP phones or the phones on the PSTN? When you say sent from the server side, is the server side the Asterisk or the Cisco? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DTMF issues/redial tones with rfc2833
Hi, I got the captured packet traces and we could see that it was coming from our asterisk server. Is there any other things that I need to look into, also we have updated from Zaptel to Dahdi 2.0.2.2, but no luck, still the random redial dtmf tones are coming in between calls...Can anyone share their opinion on this...Thank you. Regards Sandesh On Thu, Jul 8, 2010 at 5:21 PM, das sandesh sandesh...@gmail.com wrote: Thanks Zeeshan.that server is located at the headquaters and phones are at different locations, even with default rfc2833 mode, other party IVR prompts was not able to detect the tones, also 'Info' works good but not with internal options like voicemail, etc. And inband is not being used as we are using few g729 calls..Origination source of incoming calls would be from outside numbers.and we have one non sip device FXS router that handles the fax, but its not related to the voice packets... On Thu, Jul 8, 2010 at 3:42 PM, Zeeshan Zakaria zisha...@gmail.comwrote: From what you explained, it seems obvious that there exists some non-SIP device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, das sandesh sandesh...@gmail.com wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can any one share your thoughts on this, also asterisk version should not be a problem as we have other servers with same version and dtmf work good...Aslo since we also use g729 for some extensions we did not inband Also recently we got one more issue in this server, that as we talk on the phone randomly we get redial dtmf tones during the conversation, this suddenly started happening as this was good few months backI tried researching but could not find any ideas in regards to why this tones are coming into picture..I really appreciate if anyone can share their thoughts in regards to this.. Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DTMF issues/redial tones with rfc2833
Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can any one share your thoughts on this, also asterisk version should not be a problem as we have other servers with same version and dtmf work good...Aslo since we also use g729 for some extensions we did not inband Also recently we got one more issue in this server, that as we talk on the phone randomly we get redial dtmf tones during the conversation, this suddenly started happening as this was good few months backI tried researching but could not find any ideas in regards to why this tones are coming into picture..I really appreciate if anyone can share their thoughts in regards to this.. Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DTMF issues/redial tones with rfc2833
Thanks Zeeshan.that server is located at the headquaters and phones are at different locations, even with default rfc2833 mode, other party IVR prompts was not able to detect the tones, also 'Info' works good but not with internal options like voicemail, etc. And inband is not being used as we are using few g729 calls..Origination source of incoming calls would be from outside numbers.and we have one non sip device FXS router that handles the fax, but its not related to the voice packets... On Thu, Jul 8, 2010 at 3:42 PM, Zeeshan Zakaria zisha...@gmail.com wrote: From what you explained, it seems obvious that there exists some non-SIP device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, das sandesh sandesh...@gmail.com wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can any one share your thoughts on this, also asterisk version should not be a problem as we have other servers with same version and dtmf work good...Aslo since we also use g729 for some extensions we did not inband Also recently we got one more issue in this server, that as we talk on the phone randomly we get redial dtmf tones during the conversation, this suddenly started happening as this was good few months backI tried researching but could not find any ideas in regards to why this tones are coming into picture..I really appreciate if anyone can share their thoughts in regards to this.. Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Call drops on group paging asterisk - 1.4.22.1
Thanks Mike! We are using one Aastra phone with expansion module and the remaining 27 phones are from Yealink (new phones that came out), currently Aastra phone used to freeze while paging, but now we replaced the aastra to Yealink and will see if this solves the problem. Sandesh On Fri, Jun 25, 2010 at 12:02 PM, Mike l...@net-wall.com wrote: The phone brand and model might matter here, I have had no such problems with Polycom phones. Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *das sandesh *Sent:* Friday, June 25, 2010 12:58 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call drops on group paging asterisk - 1.4.22.1 Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Call drops on group paging asterisk - 1.4.22.1
Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 log problem
Thanks Dean for the information,
Currently we have the log rotate which removes the log older than 5 days.
Thanks
Sandesh
On Fri, Jun 11, 2010 at 11:19 AM, Dean Hoover dhoo...@centonline.comwrote:
Sandesh,
I'm running 1.4.23.1 right now, created this script for the logs, and
set it to run as a cron every hour. Perhaps you can modify it to help you.
#3
#
# asterisk_log_rotate.sh
#
# Removes all log files older than 7 days, then
# rotates the asterisk debug logs.
#
/usr/bin/find /var/log/asterisk -maxdepth 1 -type f -mtime +7 -exec rm
-rf {} \;
/usr/sbin/asterisk -rx logger rotate
exit 0
--
Dean Hoover
On 6/11/2010 11:08 AM, das sandesh wrote:
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would
be around 322 concurrent calls going on, but I can see that full log
grows rapidly, in one day it reaches to around 10-15 GB if I turn on the
sip debug and its tedious even by using any commands to get the required
call from the log if there is any problem. Is there any way of splitting
the full log into parts through out the day if its reaching 1GB or 2GB,
so that we can access it?
Thanks so much for your suggestions.
Regards
Sandesh
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[asterisk-users] asterisk log problem
Hi All, We have built an asterisk server (asterisk - 1.4.26.2) where there would be around 322 concurrent calls going on, but I can see that full log grows rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug and its tedious even by using any commands to get the required call from the log if there is any problem. Is there any way of splitting the full log into parts through out the day if its reaching 1GB or 2GB, so that we can access it? Thanks so much for your suggestions. Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Call Drops while doing assisted transfer from remote location
Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the second senario, if the same call comes to the main office and if they try to do an assisted transfer it works properly, but only from remote office phones connected at their homes is not working as calls are lost. Can anyone input some ideas on why it could happen, as even in the logs it does'nt give much information... We are using Yealink T28 phones. Asterisk version: 1.4.21.2, dahdi: 2.2.0.2 Thank you so much for your help. Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Regarding - P-Asserted identity and Privacy - SOLVED
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten = _1NX,n,Set(PRIVACY=${SIP_HEADER(Privacy)})
exten = _1NX,n,ExecIf($[${PRIVACY} = id]|SetCallerPres|prohib)
This makes the calls with privay ON sent as anonymous at the other end. One
more thing is to make sure you enable usecallingpres=yes in chan_dahdi.conf.
Thank you
Sandesh
On Fri, Mar 5, 2010 at 11:18 AM, das sandesh sandesh...@gmail.com wrote:
Hi All,
We have two servers, one server (SIP asterisk server) sending calls to the
second server(has PRI) which goes our through the PRI's (using TE 412p).
When the pprivacy is enabled: P-Asserted-Identity Header, privacy id are
sent in the header of SIP invite packet to the second server, how can we
identify this privacy and block the callerid as the call goes to the second
server which has the PRI cards (TDM circuit)? I tried setCallerPres(prob)
but it prohibits all calls, is there any way of identifying the calls with
the privacy ON coming from the first server and then block only those calls?
Server details:asterisk: 1.4.26.2
dahdi: 2.2.0.2
libpri: 1.4.10.1
Thanks for your help.
Regards
Sandesh
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[asterisk-users] Regarding - P-Asserted identity
Hi All, We have two servers, one server (SIP asterisk server) sending calls to the second server(has PRI) which goes our through the PRI's (using TE 412p). When the pprivacy is enabled: P-Asserted-Identity Header, privacy id are sent in the header of SIP invite packet to the second server, how can we identify this privacy and block the callerid as the call goes to the second server which has the PRI cards (TDM circuit)? I tried setCallerPres(prob) but it prohibits all calls, is there any way of identifying the calls with the privacy ON coming from the first server and then block only those calls? Server details:asterisk: 1.4.26.2 dahdi: 2.2.0.2 libpri: 1.4.10.1 Thanks for your help. Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
Hi Leif, Thanks for the information. I checked the /tmp/ folder and there was core files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes. Regards Sandesh On Thu, Feb 11, 2010 at 6:29 AM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: das sandesh wrote: Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting: But I am trying to find why did it stopped (and there was no record of asterisk stopped?) and then get restarted.In the log I could also see : [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:08] VERBOSE[7004] logger.c: -- Remote UNIX connection [Feb 10 08:02:08] VERBOSE[28272] logger.c: -- Remote UNIX connection disconnected During this period (from 8:02 till 8:29) all the phones went to 'No Service'I checked all the logs and could not find any reason why it was down or any log that shows asterisk was down at that point..any ideas are appreciated... Check /tmp/ as there may be a core.# file there which you could generate a backtrace from to determine the issue (if you're able to understand what is outputs :)) At the least, if such a core file exists, then you could line it up with the time to see if that was indeed the case. If there is a core file, it means something caused Asterisk to crash. Asterisk version: 1.4.18.1 This version is quite old, and if the issue was a crash that brought the system down momentarily, it is possible this issue may already be resolved. Leif! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk sudden restart - 1.4.18.1
Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting: But I am trying to find why did it stopped (and there was no record of asterisk stopped?) and then get restarted.In the log I could also see : [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:08] VERBOSE[7004] logger.c: -- Remote UNIX connection [Feb 10 08:02:08] VERBOSE[28272] logger.c: -- Remote UNIX connection disconnected During this period (from 8:02 till 8:29) all the phones went to 'No Service'I checked all the logs and could not find any reason why it was down or any log that shows asterisk was down at that point..any ideas are appreciated... Asterisk version: 1.4.18.1 Thanks so much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold
Thanks for all your inputs... Jeff: I used the converter to convert the sound file to 8KHz, 16 bit PCM encoded mono wave file which asterisk needs and now the POTS call quality is lot clear than before but the cell phone is still the same, not much clear...i think because of its voice codec as you mentioned. Regards Sandesh On Sat, Jan 30, 2010 at 10:38 AM, hin lee hi...@yahoo.com wrote: I am also having this issue with the MOH. Would be nice to find a solution! -- *From:* Steve Edwards asterisk@sedwards.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Fri, January 29, 2010 3:43:12 PM *Subject:* Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to process more calls. Are you in a severely storage challenged environment? You should store all of your audio encoded to match the codec used by the channel. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help for MOH - sounding scratchy/static on hold
Hi All, I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make this music on hold clear when using music files? (Where as the commercial that we record from the phone and use it as message on hold then its clear when the call is on hold, since its recording is compatible with asterisk: 8000Hz, 16 bits PCM encoded). My versions of asterisk: 1.4.18.1. I appreciate your advices. Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling SIP error codes/ISDN codes
Hi,
I was trying to use 2 of asterisk servers and interconnected, one of them as
a peer to other sever (configured in sip.conf), so all the calls to server 1
will just be passed to server 2 (has PRI Card, TE 412P, only one PRI
connected), i was sending calls to server 1 and that would send to server 2
and then dial out using Dahdi, but the problem that i got was the hangup
cause codes, i was not able to pass the appropriate ones to the server 0
(Test server) that sent a call to server 1. For example, when the user at
server 0 (test server) made a call to server 1, that sends it to server 2
and connects to the appropriate destination, but in the mean while if we
just cancel the call, we need to see the SIP error code as 487 - Request
terminated, but I was only able to see the ISDN core in PRI debug on server
2, but was not able to see '487' in sip debug, even though if i am handling
the error code conditionsIs there any way of handling the error
codes properly?
Asterisk version: 1.4.22.1
Libpri: 1.4.10.1
dahdi: 2.2.0.2 are the versions that I am using.
The way I was handling the codes for the server 2:
[macro-result]
exten = s,1,Wait(1)
exten = s,2,ResetCDR(w)
exten = s,3,NoCDR()
exten = s,4,GotoIf($[${ISNULL(${ARG1})}]?7:5)
exten = s,5,Set(RC=${ARG1})
exten = s,6,Goto(s|9)
exten = s,7,GotoIf($[${ISNULL(${DIALSTATUS})}]?8:rc-${DIALSTATUS}|1)
exten = s,8,Set(RC=${IF($[${ISNULL(${HANGUPCAUSE})}]?0:${HANGUPCAUSE})})
exten = s,9,Goto(rc-${RC}|1)
exten = s,10,Hangup(${RC})
exten = i,1,Set(RC=0)
exten = i,2,Goto(s|9)
exten = rc-ANSWER,1,Set(RC=16)
exten = rc-ANSWER,2,Goto(s|9)
exten = rc-BUSY,1,Set(RC=17)
exten = rc-BUSY,2,Goto(s|9)
exten = rc-CANCEL,1,Set(RC=16)
exten = rc-CANCEL,2,Goto(s|9)
exten = rc-CHANUNAVAIL,1,Set(RC=44)
exten = rc-CHANUNAVAIL,2,Goto(s|9)
exten = rc-CONGESTION ,1,Set(RC=19)
exten = rc-CONGESTION ,2,Goto(s|9)
;exten = rc-NOANSWER,1,Set(RC=19)
;exten = rc-NOANSWER,2,Goto(s|9)
exten = rc-0,1,NoOp(NOTDEFINED)
exten = rc-0,n,Goto(s|10)
exten = rc-1,1,NoOp(UNALLOCATED)
exten = rc-1,n,Goto(s|10)
exten = rc-2,1,NoOp(NO_ROUTE_TRANSIT_NET)
exten = rc-2,n,Goto(s|10)
exten = rc-3,1,NoOp(NO_ROUTE_DESTINATION)
exten = rc-3,n,Goto(s|10)
exten = rc-6,1,NoOp(CHANNEL_UNACCEPTABLE)
exten = rc-6,n,Goto(s|10)
exten = rc-7,1,NoOp(CALL_AWARDED_DELIVERED)
exten = rc-7,n,Goto(s|10)
exten = rc-16,1,NoOp(NORMAL_CLEARING)
exten = rc-16,n,Goto(s|10)
exten = rc-17,1,NoOp(USER_BUSY)
;exten = rc-17,n,Busy()
exten = rc-17,n,Goto(s|10)
exten = rc-18,1,NoOp(NO_USER_RESPONSE)
exten = rc-18,n,Goto(s|10)
exten = rc-19,1,NoOp(NO_ANSWER)
exten = rc-19,n,Goto(s|10)
exten = rc-21,1,NoOp(CALL_REJECTED)
exten = rc-21,n,Goto(s|10)
exten = rc-28,1,NoOp(INVALID_NUMBER_FORMAT)
exten = rc-28,n,Goto(s|10)
exten = rc-29,1,NoOp(FACILITY_REJECTED)
exten = rc-29,n,Goto(s|10)
exten = rc-30,1,NoOp(RESPONSE_TO_STATUS_ENQUIRY)
exten = rc-30,n,Goto(s|10)
exten = rc-31,1,NoOp(NORMAL_UNSPECIFIED)
exten = rc-31,n,Goto(s|10)
exten = rc-34,1,NoOp(NORMAL_CIRCUIT_CONGESTION)
exten = rc-34,n,Congestion()
exten = rc-34,n,Goto(s|10)
Thank you for your help.
Regards
Sandesh
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[asterisk-users] Question regarding digital card TE412p
Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding digital card TE412p
Thanks Victor and Vinícius for the information. I will not be doing any transcoding but using some AGI scripts, I will update the status once I configure and start using them. Thanks Sandesh On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor christ...@victormedia.dewrote: Hi! Having two TE410P with heavy load in a Pentium4 3,2GHz system running Asterisk 1.2 was no problem. It did only IVR and bridging with no transcoding though. Chris 2009/12/14 das sandesh sandesh...@gmail.com: Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring group issue
Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
Hi Alex, I am using Ring All channel strategy... Thanks Sandesh On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov abalas...@evaristesys.comwrote: What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
We are using SIP channel technology... On Tue, Nov 24, 2009 at 11:03 AM, Alex Balashov abalas...@evaristesys.comwrote: I am talking about the endpoints (extensions). SIP? DAHDI? IAX? H.323? das sandesh wrote: Hi Alex, I am using Ring All channel strategy... Thanks Sandesh On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Thanks Dave, I added a new SSD harddrive instead of a normal SATA harddrive as well as included my ip in the hosts file also I have included 'skip-name-resolv' as you mentioned to not to resolv and tested for around 250 concurrent calls, connection was going through fine...next week I would be testing it for some more additional calls..Thanks for all your replies. Best Regards Sandesh On Fri, Nov 6, 2009 at 8:01 AM, David Gibbons d...@videon-central.comwrote: I've seen asterisk really bog in internal networks if the mysql server has name resolution turned on (dns issues of course). The query will be blocked until the name resolution times out. Try adding this line the [mysqld] section of your my.cnf: [mysqld] skip-name-resolv That sent a server from ~10 seconds/query down to milliseconds. --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP 503 instead of SIP 480 in asterisk debug mode
Hi All,
I was actually trying to use the dialplan application that uses 'Dial' and
when the: Dial(SIP/xxx...@|20|) command is executed and the
destination number rings for 20 sec after which I receive as 503 Service
Unavailable, but not 480 Temporarily unavailable.
Dial(SIP/xxx...@|20|)
exten = XX,n,NoOp(Dialstatus:${DIALSTATUS})
exten = XX,n,Congestion
I can see that the DialStatus is NoAnswer but sends the 503 Service
unavailable message instead of 480 Temporarily Unavailable. Is there any
way of trying to get as 480 Temporarily available as this is the industry
standard for 'NoANSWER' ?
Thank you very much for your help.
Best Regards
Sandesh
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Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Thanks for all the information. Benaiad: I will try adding in the hosts file and try it once again, also one more that I was in regards to the harddrive, so I thought of replacing with a SSD with high read and write speeds just to check whether its going to reduce the dealy... Regards Sandesh On Thu, Oct 22, 2009 at 6:20 PM, Benaiad bena...@gmail.com wrote: Abdulmnem Benaiad Almontaha CTO Almontaha IT Co. cell: +218 92 5200025 fax: +218 21 4835263 www.almontaha.com.ly On Wed, Oct 21, 2009 at 11:57 PM, das sandesh sandesh...@gmail.comwrote: Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? Thanks Sandesh On Wed, Oct 21, 2009 at 3:57 PM, Matt Riddell li...@venturevoip.comwrote: On 22/10/09 7:30 AM, das sandesh wrote: Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: Without knowing what you're optimising you're unlikely to have much luck just setting values. We have had quite good success with the tunish-primer.sh script: http://www.day32.com/MySQL/ http://www.day32.com/MySQL/tuning-primer.sh We run with MySQL at about 500 queries per second with no problems - we don't however use Asterisk's MySQL libraries. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) Regarding Mysql delay, I've faced this problem before in my local network and this problem has been solved by adding all of my servers IP's and names in the /etc/hosts file. even when connecting locally, I think , asterisk will use a real IP instead of 127.0.0.1 when connecting to mysql and mysql will try to resolve it's name, and this step will take some time. Regards. -- Benaiad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 capacity calls using asterisk
Thanks for the information, I will look into both cisco and adtran see which would be helpful On Thu, Oct 15, 2009 at 4:09 PM, Alex Balashov abalas...@evaristesys.comwrote: David Backeberg wrote: On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote: There's no one-step solution I'm aware of. Cisco sells something called an AS5300 that supposedly can terminate a DS3 and convert it all to SIP. Otherwise, you need a channel bank like the Adtran MX2800 I was close, but incorrect. Cisco sells the 5XXX series, but I think the AS5300 has a lower capacity that a full DS3. The 58xx series claims to terminate multiple DS3s. I've never played with anything nicer than a Cisco 3845, which maxes out at 24T1s, just shy of what you can get out of the Adtran MX 2800. Yes, the AS5300 chassis can only do 4 T1s. You're looking for an AS5400, or another big router chassis that can take a DS3 adaptor and VFCs (like a 7200). -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Concurrent calls including mysql taking lot of time for execution
Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120 connect_timeout=80 table_cache=1024 thread_concurrency=8 long_query_time=10 tmp_table_size=64M join_buffer_size=1M thread_cache_size=200 key_buffer=32M table_cache=1024 sort_buffer_size=2M read_buffer_size=2M read_rnd_buffer_size=4M And I am running on asterisk 1.4.22.1, Quadcore processor 2.4Ghz, 4GB RAM, mysql 5.0. Some times we get dead air even after 50-100 calls. Is there any other additional parameters or variables or resources (hardware) to be looked into to increase the speed of mysql connections? Your advice is really appreciated. Thanks Sandesh. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Hi Steve, Thanks for your reply. I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? Regards Sandesh. On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? Thanks Sandesh On Wed, Oct 21, 2009 at 3:57 PM, Matt Riddell li...@venturevoip.com wrote: On 22/10/09 7:30 AM, das sandesh wrote: Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: Without knowing what you're optimising you're unlikely to have much luck just setting values. We have had quite good success with the tunish-primer.sh script: http://www.day32.com/MySQL/ http://www.day32.com/MySQL/tuning-primer.sh We run with MySQL at about 500 queries per second with no problems - we don't however use Asterisk's MySQL libraries. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
There were 2 problems that we faced, one was at around 50 calls, few calls were just dead air, and when I saw the logs I could see that it was sent to the sip provider and after that there was no log for that particular call that was having dead air, but at around 200 to 250, we could see that MySQL(Connect connid ipaddr uname pwd db) statement took around 5-10 sec to connect to the database and then the 2 queries in that code got executed pretty fast (1-2sec), and so here we had the dead air untill the call got connected (after 5-10sec). We also monitored the processor usage and it was around 15-20% CPU and memory was around 300M to 400M, so we concluded that it was not the hardware issue.based on all of your opinions i will try to see whether I can use any other language and try to do those operations.Thanks for all of your information! On Wed, Oct 21, 2009 at 10:51 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote: OK, but how do write the C program -- the Perl and php agis have defined functions for the agi commands, how do you do this in c? The same way. All languages need a library. Either you find a library that talks AGI or you write one. I wrote mine because when I started writing AGIs about 5 years ago, I didn't have much luck finding one. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outbound calls
You have to check and verify the SIP trunk details, as ext to ext works once the pbx is up, but to call out, it should go through your provider.so just recheck your provider's details. Regards Sandesh On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose sixfourimp...@hotmail.com wrote: here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message the call cannot be completed as dialed. if i call another ext it works. I posted the debug for both calls. ==outbound call=== --- Transmitting (NAT) to 10.0.0.46:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=9d9e3944ba To: 93214545 sip:93214...@10.0.0.8 sip%3a93214...@10.0.0.8 ;tag=as290bd498 Call-ID: 401d30b0a1893e80 CSeq: 13401 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:99676...@10.0.0.8 sip%3a99676...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv = ext to ext=== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=d729237fcc To: 111 sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=as553ab5e9 Call-ID: c7cc32657c620790 CSeq: 8007 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@10.0.0.8 sip%3a...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 10414 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DS3 capacity calls using asterisk
Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got stuck at 150 calls, above that not working in stress test
Hi Matt, Thanks so much for your help. I tried lot of ways to trouble shoot the issue, but finally I figured out that it was from the carrier side that they had set the limit of 150. Till now I under the impression that they provide just the bandwidth for the trunk, but they have the ability to limit the concurrent calls. Thanks Sandesh On Sun, Oct 4, 2009 at 9:06 PM, Matt Riddell li...@venturevoip.com wrote: On 3/10/09 3:55 AM, das sandesh wrote: I am using the command: ./sipp -sn uac -d 200 -s repective context pattern IP Address -l 200 Its 10 calls per second and 200 concurrent calls, similarly I used 2 ssh sessions each sending 100 concurrent calls. But this was limiting to only 150 calls. Start with 5 calls per second. Also, I don't notice anything to set it to this, are you sure you're not trying to start all those calls concurrently? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got stuck at 150 calls, above that not working in stress test
These calls are from asterisk. I am using sipp to generate the calls and I increased the limits using the command line interface and used ulimit -n 1 and as well as changed in /etc/security/limits.conf. I dint find any errors on the console... Thanks Sandesh On Thu, Oct 1, 2009 at 4:29 PM, Matt Riddell li...@venturevoip.com wrote: On 2/10/09 12:41 AM, das sandesh wrote: Hi Matt, When I get can more that 150 calls, i get a busy signal (Congestion) for the calls above 150 - says your call cannot be completed now, its allowing only 150 callsIs there any thing related to field descriptors from linux point of view that I need to increase inorder to increase the call capacity. Is that coming from Asterisk? It seems strange that Asterisk would reject the call unless you have settings in asterisk.conf to do this. You've said you've already increased the file descriptor limits - did you do this in the console you were using to subsequently run Asterisk from? Do you get any errors in the Asterisk console? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got stuck at 150 calls, above that not working in stress test
I am using the command: ./sipp -sn uac -d 200 -s repective context pattern IP Address -l 200 Its 10 calls per second and 200 concurrent calls, similarly I used 2 ssh sessions each sending 100 concurrent calls. But this was limiting to only 150 calls. Thanks Sandesh On Fri, Oct 2, 2009 at 9:23 AM, Matt Riddell li...@venturevoip.com wrote: On 3/10/09 2:40 AM, das sandesh wrote: These calls are from asterisk. I am using sipp to generate the calls and I increased the limits using the command line interface and used ulimit -n 1 and as well as changed in /etc/security/limits.conf. I dint find any errors on the console... How are you spacing the calls out? I.E. how many calls per second? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got stuck at 150 calls, above that not working in stress test
Hi Matt, When I get can more that 150 calls, i get a busy signal (Congestion) for the calls above 150 - says your call cannot be completed now, its allowing only 150 callsIs there any thing related to field descriptors from linux point of view that I need to increase inorder to increase the call capacity. Thanks Sandesh On Thu, Oct 1, 2009 at 4:19 AM, Matt Riddell li...@venturevoip.com wrote: On 1/10/09 5:56 PM, das sandesh wrote: Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the other calls are giving busy, I tried to do ulimit related stuff, like increasing the soft and hard limits to 10 but no luck, Any ideas or views are really appreciated. Also I even changed the call limit to 500, but stills it can handle only 150 total. What do you mean it handles only 150? What happens when you get above that number? I regularly have more than that active on a machine. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] got stuck at 150 calls, above that not working in stress test
Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the other calls are giving busy, I tried to do ulimit related stuff, like increasing the soft and hard limits to 10 but no luck, Any ideas or views are really appreciated. Also I even changed the call limit to 500, but stills it can handle only 150 total. Thanks for your help. Regards Sandesh. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels got stuck in asterisk 1.4.18.1
Thanks Darrick, I will try to upgrade the version. Also I got to know that if we are going to limit the call length as well as silence detection might me useful for eradicating the channel lockup. On Fri, Sep 18, 2009 at 11:52 PM, Darrick Hartman dhart...@djhsolutions.com wrote: das sandesh wrote: Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do soft hangup channel, it says Requested for soft hangup for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to kill the asterisk process and then start asterisk to come back to normal. I wanted to know did any one faced such a problem? Is there any way of getting to know if the channel gets stuck (since in our senario we came to know since the person at the extension(channel) that got stuck was not able to receive calls) or is there a way to eradicate the channel getting stuck? Thank you very much. Regards Sandesh Upgrade to a recent version of Asterisk. 1.4.26.2 is the latest 1.4 release. Not much chance you're going to get help when you're using something as 1.4.18.1. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channels got stuck in asterisk 1.4.18.1
Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do soft hangup channel, it says Requested for soft hangup for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to kill the asterisk process and then start asterisk to come back to normal. I wanted to know did any one faced such a problem? Is there any way of getting to know if the channel gets stuck (since in our senario we came to know since the person at the extension(channel) that got stuck was not able to receive calls) or is there a way to eradicate the channel getting stuck? Thank you very much. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channels stuck up even without use
Hi All, I had a problem yesterday, that our asterisk server showed 2 channels were in use continuously, but nobody were using any of them at that time. I had to kill them using softhangup and I checked all the logs but could not find why exactly this problem occurred, as the system was running fine for many months till yesterday...Once I killed the channels it is fine now, but I wanted to know where can I find what might have caused this problem? We use Polycom 320 phones and asterisk 1.4.21.2 version Thanks for the help Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels stuck up even without use
Thanks Steve. The channels which got stuck up are the SIP channels. If I use rtptimeout then if there is some silence at both the ends the call may get disconnected after that period of time also if I use Absolutetimeout then there are chances that some calls may be upto 30min to 40min, if our timeout is 1200, call will be disconnected after 20 min, so what would be the better option for this cause of SIP channels stuck up Thanks Sandesh On Thu, Aug 13, 2009 at 3:32 AM, Steve Howes st...@geekinter.net wrote: On 13 Aug 2009, at 07:51, das sandesh wrote: Hi All, I had a problem yesterday, that our asterisk server showed 2 channels were in use continuously, but nobody were using any of them at that time. I had to kill them using softhangup and I checked all the logs but could not find why exactly this problem occurred, as the system was running fine for many months till yesterday...Once I killed the channels it is fine now, but I wanted to know where can I find what might have caused this problem? We use Polycom 320 phones and asterisk 1.4.21.2 version Analogue? ISDN? SIP? If its SIP its probably a lost packet. Sticking on max call length and silence detection might help. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple contexts for multiple locations
-> [asterisk-users] multiple contexts for multiple locations asterisk-users -- Thread -- -- Date -- [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to [asterisk-users] multiple contexts for multiple locations das sandesh Re: [asterisk-users] multiple contexts for multiple locations John A. Sullivan III Reply via email to
